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Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
# Visibility considerations:
#
# Most targets in this file should have visibility ":*", as they are only
# used internally.
# Some functions are cleared for wider webrtc usage; these have default
# visibility (set to "//*", not the gn default of "*").
# These are:
# - rtc_pc
# - session_description
# - simulcast_description
# - sdp_utils
# - media_stream_observer
# - video_track_source
# - libjingle_peerconnection
#
# Some targets are depended on by external users for historical reasons,
# and are therefore marked with visibility "*". This is in the process
# of being removed.
#
# Some targets are only publicly visible in Chrome builds.
# These are marked up as such.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../webrtc.gni")
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
build: add options to configure libsrtp for boringssl or other libraries Depends on https://webrtc-review.googlesource.com/c/src/+/359928 BUG=webrtc:42234521,webrtc:42224104 Change-Id: I0d6335aa5fb3f090c781bed234ed34d6c98ec299 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359928 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42857}
2024-08-22 11:52:33 -07:00
if (rtc_build_libsrtp) {
import("//third_party/libsrtp/options.gni")
assert(rtc_build_ssl == libsrtp_build_boringssl,
"Mismatch ssl build settings detected")
assert(rtc_ssl_root == libsrtp_ssl_root, "Mismatch in ssl root detected")
}
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
group("pc") {
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
deps = [ ":rtc_pc" ]
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
}
Replace proxy ScopedEvent with TRACE_EVENT TRACE_EVENT is already scoped! #rtc_fixit Tested: Compiled the patch in Chromium and confirmed the Proxy events are still present. I can send the resulting trace to reviewers if desired. Bug: webrtc:15867 Change-Id: I5717a85c0ee25e8e20123afa08064c9b6666ba96 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342920 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/main@{#41916}
2024-03-13 12:31:11 +00:00
rtc_source_set("proxy") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Replace proxy ScopedEvent with TRACE_EVENT TRACE_EVENT is already scoped! #rtc_fixit Tested: Compiled the patch in Chromium and confirmed the Proxy events are still present. I can send the resulting trace to reviewers if desired. Bug: webrtc:15867 Change-Id: I5717a85c0ee25e8e20123afa08064c9b6666ba96 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342920 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/main@{#41916}
2024-03-13 12:31:11 +00:00
sources = [ "proxy.h" ]
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
deps = [
Apply include-cleaner to api direct files Bug: webrtc:42226242 Change-Id: Ia1e6021fc18a30b6da9b4a43118167b6ae173717 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360680 Commit-Queue: Dor Hen <dorhen@meta.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42993}
2024-09-04 14:55:22 +03:00
"../api:make_ref_counted",
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
"../api:scoped_refptr",
"../api/task_queue",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
"../rtc_base:threading",
"../rtc_base/system:rtc_export",
]
}
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("channel") {
Move pc/channel.h to only be used in .cc files This is an implementation API, user classes should in principle only use the channel_interface.h Bug: webrtc:13931 Change-Id: I85c285217858dc087c90a50792e980f731f4439f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36674}
2022-04-27 12:15:49 +00:00
visibility = [
":*",
"../test/peer_scenario",
]
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
sources = [
"channel.cc",
"channel.h",
]
deps = [
":channel_interface",
":rtp_media_utils",
":rtp_transport_internal",
":session_description",
"../api:libjingle_peerconnection_api",
Set voice RTCP mode based on the RemoteContent and not based on the LocalContent. The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content). Bug: webrtc:340041654 Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#43449}
2024-11-22 13:45:27 +00:00
"../api:rtp_headers",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/task_queue",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../api/units:timestamp",
"../call:rtp_interfaces",
"../call:rtp_receiver",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
"../media:media_channel_impl",
"../media:rid_description",
"../media:rtp_utils",
"../media:stream_params",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Clean up p2p:rtc_p2p target removing the webrtc need for having sources in it. BUG=webrtc:42226155 Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43540}
2024-12-10 09:26:57 -08:00
"../p2p:basic_packet_socket_factory",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:socket",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:threading",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base/containers:flat_set",
"../rtc_base/network:sent_packet",
"../rtc_base/third_party/sigslot",
Perform packetization verification until a match is found If there happens to be an asymmetry between local and remote codecs we shouldn't validate that there's a 1:1 packetization mapping for every codec. It's sufficient to check that there's at least one matching packetization per codec. Bug: webrtc:15473 Change-Id: Ib4fc8fdd54bb4dccf96f0c802746c848e2deed83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320440 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Auto-Submit: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40760}
2023-09-15 16:33:03 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("channel_interface") {
visibility = [ ":*" ]
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
sources = [ "channel_interface.h" ]
deps = [
":rtp_transport_internal",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("dtls_srtp_transport") {
visibility = [ ":*" ]
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
sources = [
"dtls_srtp_transport.cc",
"dtls_srtp_transport.h",
]
deps = [
":srtp_transport",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../api:field_trials_view",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("dtls_transport") {
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
visibility = [
":*",
"../test/*",
]
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
sources = [
"dtls_transport.cc",
"dtls_transport.h",
]
deps = [
":ice_transport",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
"../api:ice_transport_interface",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
"../api:libjingle_peerconnection_api",
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219}
2022-06-14 15:48:26 +02:00
"../api:make_ref_counted",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
"../api:scoped_refptr",
"../api:sequence_checker",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport",
"../p2p:dtls_transport_internal",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("external_hmac") {
visibility = [ ":*" ]
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
sources = [
"external_hmac.cc",
"external_hmac.h",
]
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
deps = [
"../rtc_base:logging",
Move zero_memory out of rtc_base_approved Bug: webrtc:9838 Change-Id: I2d1507354abf208947ac299ab41abe3cfa2b4274 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258767 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36585}
2022-04-14 13:23:51 +02:00
"../rtc_base:zero_memory",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
]
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("ice_transport") {
visibility = [ ":*" ]
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
sources = [
"ice_transport.cc",
"ice_transport.h",
]
deps = [
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
"../api:libjingle_peerconnection_api",
"../api:sequence_checker",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:threading",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("jsep_transport") {
visibility = [ ":*" ]
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
sources = [
"jsep_transport.cc",
"jsep_transport.h",
]
deps = [
":dtls_srtp_transport",
":dtls_transport",
":rtcp_mux_filter",
":rtp_transport",
":rtp_transport_internal",
":sctp_transport",
":session_description",
":srtp_transport",
":transport_stats",
"../api:array_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:ice_transport_interface",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:scoped_refptr",
Remove //api:sequence_checker from public deps Bug: webrtc:8603 Change-Id: Ib572cdc5b5f4c1f0eb3357e5e0812fc9ce7437c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257915 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36465}
2022-04-04 17:18:21 +02:00
"../api:sequence_checker",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../api/transport:datagram_transport_interface",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../call:payload_type_picker",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../media:rtc_data_sctp_transport_internal",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport",
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
"../rtc_base:threading",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("jsep_transport_collection") {
visibility = [ ":*" ]
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
sources = [
"jsep_transport_collection.cc",
"jsep_transport_collection.h",
]
deps = [
":jsep_transport",
":session_description",
"../api:libjingle_peerconnection_api",
"../api:sequence_checker",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:p2p_constants",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("jsep_transport_controller") {
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
visibility = [
":*",
"../test/peer_scenario:*",
]
sources = [
"jsep_transport_controller.cc",
"jsep_transport_controller.h",
]
deps = [
":channel",
":dtls_srtp_transport",
":dtls_transport",
":jsep_transport",
":jsep_transport_collection",
":rtp_transport",
":rtp_transport_internal",
":sctp_transport",
":session_description",
":srtp_transport",
":transport_stats",
"../api:async_dns_resolver",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:dtls_transport_interface",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../api:ice_transport_factory",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
sctp: Pass webrtc::Environment to DcSctpTransport The DcSctpTransport will soon use field trials to conditionally enable some options. And overall, there is a migration project to start using the Environment and this CL is in that direction, also setting the boundary; The dcSCTP library should not depend on it. But the transport is allowed to. Bug: webrtc:14997 Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41872}
2024-02-28 12:16:15 +01:00
"../api/environment",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../api/rtc_event_log",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/transport:sctp_transport_factory_interface",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../call:payload_type",
"../call:payload_type_picker",
"../media:codec",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../media:rtc_data_sctp_transport_internal",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport",
"../p2p:dtls_transport_factory",
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:packet_transport_internal",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../rtc_base:callback_list",
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
Ensure Call is notified of un demuxable packets With this cl, packets that are discarded in RtpTransport now notifies Call, so that they can be part of BWE even if they are dropped. These packets have been recevied on the transport, and has bin decrypted and parsed and thus can be accounted for. The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded. Bug: webrtc:14928 Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39727}
2023-03-30 16:53:59 +02:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Introduce new way to enable media in PeerConnectionFactory instead of requiring to pass in call_factory and media_engine webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41049}
2023-10-31 13:51:16 +01:00
rtc_source_set("media_factory") {
sources = [ "media_factory.h" ]
deps = [
Use Environemnt in MedaFactory::CreateMediaEngine to propagate field trials and task queue factory Bug: webrtc:15656 Change-Id: I2d19e169d2ff1cc871899a0e96b1733333fdc604 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41257}
2023-11-27 17:56:49 +01:00
"../api/environment",
Introduce new way to enable media in PeerConnectionFactory instead of requiring to pass in call_factory and media_engine webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41049}
2023-10-31 13:51:16 +01:00
"../call:call_interfaces",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Introduce new way to enable media in PeerConnectionFactory instead of requiring to pass in call_factory and media_engine webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41049}
2023-10-31 13:51:16 +01:00
]
}
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("media_session") {
rtc_pc_base breakup: Make a target visible to Chrome Bug: webrtc:13805 Change-Id: I129bf3ecb69bd7dee363f16f2b6ba39cf9d06bac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256008 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36245}
2022-03-18 09:02:59 +00:00
visibility = [ "*" ] # Used by Chrome
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
sources = [
"media_session.cc",
"media_session.h",
]
deps = [
":jsep_transport",
":media_protocol_names",
":rtp_media_utils",
":session_description",
":simulcast_description",
":used_ids",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../api:libjingle_peerconnection_api",
Include-what-you-use pc/media_session Bug: webrtc:42226242 Change-Id: I25743717d1f0e7a0305589139bd386353b4e5054 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350122 Auto-Submit: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42262}
2024-05-08 06:52:16 +00:00
"../api:rtc_error",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api/crypto:options",
Correct H.265 level-id in fmtp line for offer/answer. On a sendrecv m-line, the offered level-id represents the maximum that can be both sent and received; on a sendonly m-line, the offered level-id represents the maximum that can be sent; on a recvonly m-line, the offered level-id represents the maximum that can be received. Also according to RFC 7798 section 5, the highest level indicated by the answer is either equal to or lower than that in the offer Bug: chromium:41480904 Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> Cr-Commit-Position: refs/heads/main@{#43425}
2024-11-18 21:35:59 +08:00
"../api/video_codecs:video_codecs_api",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../call:payload_type",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rid_description",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../media:rtc_data_sctp_transport_internal",
"../media:rtc_sdp_video_format_utils",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:stream_params",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_credentials_iterator",
"../p2p:p2p_constants",
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_description_factory",
"../p2p:transport_info",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Reland "Don't create channel_manager++ when media_engine is not set" This reverts commit c6c02efb56b24df04ed9ab61252c14c7bddcca93. Reason for revert: Test now passes (and channel manager is gone) Original change's description: > Revert "Don't create channel_manager when media_engine is not set" > > This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f. > > Reason for revert: breaks downstream project > > Original change's description: > > Don't create channel_manager when media_engine is not set > > > > Also remove a bunch of functions in ChannelManager that were just > > forwarding to MediaEngineInterface. > > > > Bug: webrtc:13931 > > Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36801} > > Bug: webrtc:13931 > Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#36811} Bug: webrtc:13931 Change-Id: I7b5b45b46095c18d489b6a9fe4c625971d6b3da6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261661 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36976}
2022-05-23 14:57:47 +00:00
"../rtc_base/memory:always_valid_pointer",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("media_stream_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "media_stream_proxy.h" ]
deps = [
":proxy",
"../api:media_stream_interface",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("media_stream_track_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "media_stream_track_proxy.h" ]
deps = [
":proxy",
"../api:media_stream_interface",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("peer_connection_factory_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "peer_connection_factory_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("peer_connection_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "peer_connection_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
Add PeerConnectionInterface::ReconfigureBandwidthEstimation Using the Api, BWE components are recreated and new settings can be applied. Initially, the only configuration available is allowing BWE probes without media". Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration. Integration test in https://webrtc-review.googlesource.com/c/src/+/337322 Bug: webrtc:14928 Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41687}
2024-02-07 14:16:20 +01:00
"../api/transport:bandwidth_estimation_settings",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtcp_mux_filter") {
visibility = [ ":*" ]
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
sources = [
"rtcp_mux_filter.cc",
"rtcp_mux_filter.h",
]
deps = [
":session_description",
"../rtc_base:logging",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtp_media_utils") {
visibility = [ ":*" ]
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
sources = [
"rtp_media_utils.cc",
"rtp_media_utils.h",
]
deps = [
"../api:rtp_transceiver_direction",
"../rtc_base:checks",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtp_receiver_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "rtp_receiver_proxy.h" ]
deps = [
":proxy",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:dtls_transport_interface",
"../api:frame_transformer_interface",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
"../api:libjingle_peerconnection_api",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtp_sender_proxy") {
visibility = [ ":*" ]
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
sources = [ "rtp_sender_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtp_transport") {
visibility = [ ":*" ]
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
sources = [
"rtp_transport.cc",
"rtp_transport.h",
]
deps = [
":rtp_transport_internal",
":session_description",
"../api:array_view",
Dont signal ReadyToSend in RtpTransport::SendPacket Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN. ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail. Changing the state in the context of sending packets may cause recursive calls and seems to cause problems with incorrect states. It is simpler if RtpTransport::SendPacket ignore these failures and upper layers treat these lost packets similar to if the packets had been lost via UDP. For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/. Bug: webrtc:361124449 b/359989715 Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42868}
2024-08-22 08:54:32 +00:00
"../api:field_trials_view",
Fire MaybeSignalReadyToSend in a PostTask when recursive Speculative fix. Writing the test for it is more complex. Bug: chromium:1483874 Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40820}
2023-09-26 19:38:19 +00:00
"../api/task_queue:pending_task_safety_flag",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../api/units:timestamp",
"../call:rtp_receiver",
Split up the call/video_stream_api target The split shows that some places don't need it at all. Most other places will depend on both send and receive stream targets. Bug: webrtc:373151158 Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43230}
2024-10-13 21:08:30 +00:00
"../call:video_receive_stream_api",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rtp_utils",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base:logging",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base:socket",
Propagate rtc::ReceivedPacket further in RtpTransport Bug: webrtc:15368 Change-Id: I4c8989a7b9efacbcc29c0c3331d8f4d7350774c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341220 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41825}
2024-02-27 09:25:11 +01:00
"../rtc_base/network:received_packet",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base/network:sent_packet",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("rtp_transport_internal") {
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
visibility = [
":*",
"../test/peer_scenario",
]
sources = [ "rtp_transport_internal.h" ]
deps = [
":session_description",
"../call:rtp_receiver",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
Convert signals in rtp_transport_internal.h to CallbackList Bug: webrtc:11943 Change-Id: I8e0839363712d9d8b49c2f6cbdb5f3ac59d79219 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318882 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40700}
2023-09-05 09:49:32 +00:00
"../rtc_base:callback_list",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("sctp_transport") {
visibility = [ ":*" ]
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
sources = [
"sctp_transport.cc",
"sctp_transport.h",
]
deps = [
":dtls_transport",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../api:libjingle_peerconnection_api",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../api:rtc_error",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../api:scoped_refptr",
"../api:sequence_checker",
Remove sigslot usage from SctpTransportInternal Bug: webrtc:11943 Change-Id: I42edf8e2e15e580bcda090447a7aae4a56366b33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270661 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37867}
2022-08-22 11:34:29 +02:00
"../api/transport:datagram_transport_interface",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../media:rtc_data_sctp_transport_internal",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../rtc_base:checks",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../rtc_base:copy_on_write_buffer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("sctp_utils") {
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
visibility = [
":*",
Permit all fuzzers to use sctp_utils Bug: chromium:1306284 Change-Id: Icaa4d13e9c85cdae48a2ae575b5067fe13d805b6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255421 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36201}
2022-03-15 06:53:07 +00:00
"../test/fuzzers:*",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
]
sources = [
"sctp_utils.cc",
"sctp_utils.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api/transport:datagram_transport_interface",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Remove the `SctpDataChannel::config_` member variable. Instead there are direct member variables for the various relevant states, some weren't needed, some can be const but the `id` member in particular needs special handling and can't be const. For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special `-1` case (for what's essentially an unsigned 16 bit int). Using a special type for this also has the effect that range checking happens more consistently (although I'm not modifying the structs in api/). With upcoming steps of avoiding thread hops, the ID may need to migrate to the network thread, which the thread checks will help with. Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file to help with more accurately tracking code coverage. Bug: webrtc:11547 Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39539}
2023-03-12 16:59:25 +01:00
"../media:rtc_data_sctp_transport_internal",
"../net/dcsctp/public:types",
Move byte_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic7e912cba1218f1eed794cb8c393ac148106b16c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259310 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36634}
2022-04-19 17:35:04 +02:00
"../rtc_base:byte_buffer",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
rtc_source_set("srtp_session") {
visibility = [ ":*" ]
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
sources = [
"srtp_session.cc",
"srtp_session.h",
]
deps = [
":external_hmac",
"../api:array_view",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../api:scoped_refptr",
"../api:sequence_checker",
"../modules/rtp_rtcp:rtp_rtcp_format",
srtp: spanify key setters BUG=webrtc:357776213 Change-Id: I307085690588e324409bb32a3db5ec9cfa99df52 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362126 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43055}
2024-09-13 20:30:12 -07:00
"../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
Move global libsrtp usage count into a singleton class Avoids using webrtc::GlobalMutex. Since state is allocated on first use and never destroyed, we avoid an exit-time destructor when building with absl::Mutex. Bug: webrtc:11567 Change-Id: Ib9c6480ab0474e37a853460115b35d961b93009c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258080 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36455}
2022-04-05 16:14:21 +02:00
"../rtc_base:macromagic",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
"../rtc_base/synchronization:mutex",
"../system_wrappers:metrics",
Adopt absl::string_view in rtc_base/string_encode.* Bug: webrtc:13579 Change-Id: If52108d151a12bde0e8d552ce7940948c08cef3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256812 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36424}
2022-03-31 10:36:48 +02:00
"//third_party/abseil-cpp/absl/base:core_headers",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Adopt absl::string_view in rtc_base/string_encode.* Bug: webrtc:13579 Change-Id: If52108d151a12bde0e8d552ce7940948c08cef3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256812 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36424}
2022-03-31 10:36:48 +02:00
]
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
rtc_source_set("srtp_transport") {
visibility = [ ":*" ]
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
sources = [
"srtp_transport.cc",
"srtp_transport.h",
]
deps = [
":rtp_transport",
":srtp_session",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rtp_utils",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../rtc_base:safe_conversions",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Move zero_memory out of rtc_base_approved Bug: webrtc:9838 Change-Id: I2d1507354abf208947ac299ab41abe3cfa2b4274 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258767 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36585}
2022-04-14 13:23:51 +02:00
"../rtc_base:zero_memory",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
"//third_party/abseil-cpp/absl/strings",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("transport_stats") {
visibility = [ ":*" ]
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
sources = [
"transport_stats.cc",
"transport_stats.h",
]
deps = [
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
"../api:libjingle_peerconnection_api",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("used_ids") {
visibility = [ ":*" ]
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
sources = [ "used_ids.h" ]
deps = [
"../api:rtp_parameters",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
rtc_source_set("video_track_source_proxy") {
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
visibility = [ "*" ] # Used by Chrome
sources = [
"video_track_source_proxy.cc",
"video_track_source_proxy.h",
]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:video_track_source_constraints",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../rtc_base:threading",
]
Add empty build targets for breakup of rtc_pc_base This is an initial step, defining targets so that external users can add dependencies. Bug: webrtc:13805 Change-Id: I5d2caf2fd746d459f747c390eb783bb4d6f179ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254480 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36168}
2022-03-10 07:30:31 +00:00
}
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
rtc_source_set("session_description") {
Mark webrtc_sdp as visible target It is being used, among others, by Chrome. Bug: webrtc:13634 Change-Id: I61954e3a149ead7abf47efe67d6fbecd63ad8d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35928}
2022-02-07 08:39:52 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ "*" ] # Used by Chrome and others
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
sources = [
Break out pc/session_description build target (part 2) Move the files. Bug: webrtc:11967 Change-Id: Ib8c26e32248064ce6140bef79410e5a4a595d01e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221861 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34266}
2021-06-10 06:51:48 +00:00
"session_description.cc",
"session_description.h",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
]
deps = [
":media_protocol_names",
":simulcast_description",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
"../media:stream_params",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
"../rtc_base:checks",
"../rtc_base:socket_address",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:stringutils",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/algorithm:container",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
]
}
rtc_source_set("simulcast_description") {
sources = [
Break out pc/session_description build target (part 2) Move the files. Bug: webrtc:11967 Change-Id: Ib8c26e32248064ce6140bef79410e5a4a595d01e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221861 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34266}
2021-06-10 06:51:48 +00:00
"simulcast_description.cc",
"simulcast_description.h",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
]
deps = [
"../rtc_base:checks",
"../rtc_base:socket_address",
"../rtc_base/system:rtc_export",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
]
}
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
rtc_source_set("rtc_pc") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
if (build_with_chromium) {
visibility = [ "*" ]
}
Remove legacy video codec factories. Removes the deprecated video codec factories and the related flag and helper classes. Bug: webrtc:7925 Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98 Reviewed-on: https://webrtc-review.googlesource.com/c/113940 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26245}
2019-01-14 14:23:23 +01:00
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
Remove internal dependencies on rtc_pc_base This explores the theory that targets that have no files, just dependencies, are unnecessary. Bug: webrtc:13805 Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36363}
2022-03-29 07:53:50 +00:00
deps = [ "../media:rtc_audio_video" ]
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("media_protocol_names") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
sources = [
"media_protocol_names.cc",
"media_protocol_names.h",
]
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
deps = [ "//third_party/abseil-cpp/absl/strings:string_view" ]
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
}
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
rtc_library("sctp_data_channel") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
sources = [
"sctp_data_channel.cc",
"sctp_data_channel.h",
]
deps = [
":data_channel_utils",
":proxy",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":sctp_utils",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api:rtc_error",
"../api:scoped_refptr",
Add thread checker to SctpSidAllocator Also changing AllocateSid to return StreamId instead of bool. Bug: webrtc:11547 Change-Id: I776e917300ddfdbb79e78c01ef880209ec2c5917 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298301 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39630}
2023-03-21 18:45:24 +01:00
"../api:sequence_checker",
[DataChannelInterface] Introduce DataChannelInterface::SendAsync() One problem with the existing Send() method is that it has a return value that is problematic for a fully async implementation. A second problem with Send() is that the return value is bool and not RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError. Also, start deprecating `bool Send()` in favor of `void SendAsync()` and adding `network_safety_` flag for posting async operations to the network thread. This flag also takes over from the `connected_to_transport_` which can now be removed. Bug: webrtc:11547, webrtc:13289 Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39817}
2023-04-11 17:32:34 +02:00
"../api/task_queue:pending_task_safety_flag",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../api/transport:datagram_transport_interface",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../media:rtc_data_sctp_transport_internal",
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:logging",
"../rtc_base:macromagic",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:threading",
Use WeakPtr in SctpDataChannel. DataChannelController used WeakPtr to clear outstanding references upon destruction - except for the case of SctpDataChannel where we had a pointer+flag for the same purpose. This change updates SctpDataChannel and FakeDataChannelController to use a consistent approach. Bug: none Change-Id: I0248471c241365a2c0de76afbb37302115650194 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39464}
2023-03-02 15:42:06 +01:00
"../rtc_base:weak_ptr",
Remove the `SctpDataChannel::config_` member variable. Instead there are direct member variables for the various relevant states, some weren't needed, some can be const but the `id` member in particular needs special handling and can't be const. For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special `-1` case (for what's essentially an unsigned 16 bit int). Using a special type for this also has the effect that range checking happens more consistently (although I'm not modifying the structs in api/). With upcoming steps of avoiding thread hops, the ID may need to migrate to the network thread, which the thread checks will help with. Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file to help with more accurately tracking code coverage. Bug: webrtc:11547 Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39539}
2023-03-12 16:59:25 +01:00
"../rtc_base/containers:flat_set",
Add thread checker to SctpSidAllocator Also changing AllocateSid to return StreamId instead of bool. Bug: webrtc:11547 Change-Id: I776e917300ddfdbb79e78c01ef880209ec2c5917 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298301 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39630}
2023-03-21 18:45:24 +01:00
"../rtc_base/system:no_unique_address",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base/system:unused",
]
}
rtc_library("data_channel_utils") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
visibility = [ "*" ] # Known to be used externally
Mark webrtc_sdp as visible target It is being used, among others, by Chrome. Bug: webrtc:13634 Change-Id: I61954e3a149ead7abf47efe67d6fbecd63ad8d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35928}
2022-02-07 08:39:52 +00:00
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
sources = [
"data_channel_utils.cc",
"data_channel_utils.h",
]
deps = [
"../api:libjingle_peerconnection_api",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:checks",
]
}
Separate PC resources required at runtime from PCfactory This enables modules that share the resources to reuse the connection context object but not take a dependency on PeerConnectionFactory. Bug: webrtc:11967 Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32412}
2020-10-15 08:34:31 +00:00
rtc_library("connection_context") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Separate PC resources required at runtime from PCfactory This enables modules that share the resources to reuse the connection context object but not take a dependency on PeerConnectionFactory. Bug: webrtc:11967 Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32412}
2020-10-15 08:34:31 +00:00
sources = [
"connection_context.cc",
"connection_context.h",
]
deps = [
Introduce new way to enable media in PeerConnectionFactory instead of requiring to pass in call_factory and media_engine webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41049}
2023-10-31 13:51:16 +01:00
":media_factory",
Separate PC resources required at runtime from PCfactory This enables modules that share the resources to reuse the connection context object but not take a dependency on PeerConnectionFactory. Bug: webrtc:11967 Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32412}
2020-10-15 08:34:31 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
Simplify reference counting implementation of PendingTaskSafetyFlag. On a 32bit system, this reduces the allocation size of the flag down from 12 bytes to 8, and removes the need for a vtable (the extra 4 bytes are the vtable pointer). The downside is that this change makes the binary layout of the flag, less compatible with RefCountedObject<> based reference counting objects and thus we don't immediately get the benefits of identical COMDAT folding and subsequently there's a slight binary size increase. With wider use, the binary size benefits will come. Bug: none Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33793}
2021-04-20 16:58:01 +02:00
"../api:refcountedbase",
Separate PC resources required at runtime from PCfactory This enables modules that share the resources to reuse the connection context object but not take a dependency on PeerConnectionFactory. Bug: webrtc:11967 Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32412}
2020-10-15 08:34:31 +00:00
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Use Environment to keep peer connection factory field trials in ConnectionContext Bug: webrtc:15656 Change-Id: Ice52fcb9ba54a5d0034b59233ceae4f9cefbceae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328860 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41252}
2023-11-27 15:00:21 +01:00
"../api/environment",
Delete the "initialize" method of PeerConnectionFactory Also remove the "initialized" concept from ConnectionContext. This CL also always creates the objects on the signaling thread. Makes the initialization code slightly more readable. Bug: webrtc:11967 Change-Id: I5e451a3c5225c29c30d32bb4843df8c107ec30c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188626 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32453}
2020-10-20 15:35:31 +00:00
"../api/neteq:neteq_api",
"../api/transport:sctp_transport_factory_interface",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../media:rtc_data_sctp_transport_factory",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:basic_packet_socket_factory",
Delete the "initialize" method of PeerConnectionFactory Also remove the "initialized" concept from ConnectionContext. This CL also always creates the objects on the signaling thread. Makes the initialization code slightly more readable. Bug: webrtc:11967 Change-Id: I5e451a3c5225c29c30d32bb4843df8c107ec30c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188626 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32453}
2020-10-20 15:35:31 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network",
"../rtc_base:rtc_certificate_generator",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:socket_factory",
"../rtc_base:socket_server",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Use AlwaysValidPointer in connection_context This extends AlwaysValidPointer to take a lambda for its default rather than requesting a constructor. Bug: none Change-Id: Ied97968c3f511af15422a1eef9801d14d4ec5b96 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260580 Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36745}
2022-05-03 11:37:34 +00:00
"../rtc_base/memory:always_valid_pointer",
Separate PC resources required at runtime from PCfactory This enables modules that share the resources to reuse the connection context object but not take a dependency on PeerConnectionFactory. Bug: webrtc:11967 Change-Id: Ic68cbf061b3226f02f8638abd79ad881e89951d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188120 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32412}
2020-10-15 08:34:31 +00:00
]
}
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
rtc_source_set("data_channel_controller") {
visibility = [ ":*" ]
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
sources = [
"data_channel_controller.cc",
"data_channel_controller.h",
]
deps = [
":data_channel_utils",
":peer_connection_internal",
":sctp_data_channel",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":sctp_utils",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../api:libjingle_peerconnection_api",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
Add safety flag to DataChannelController for PostTask. Bug: none Change-Id: If70c1cd9fb4e6fefefa96df819deac052655c49a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296140 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39485}
2023-03-06 12:51:39 +01:00
"../api/task_queue:pending_task_safety_flag",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../api/transport:datagram_transport_interface",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../rtc_base:logging",
"../rtc_base:macromagic",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out data_channel_controller from peerconnection target Bug: webrtc:11943 Change-Id: I80c6b1be770f4e99cc7e21ef0e30977198166f88 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251323 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35949}
2022-02-08 10:49:09 +00:00
"../rtc_base:threading",
"../rtc_base:weak_ptr",
"//third_party/abseil-cpp/absl/algorithm:container",
]
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
}
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
rtc_source_set("peer_connection_internal") {
visibility = [ ":*" ]
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
sources = [ "peer_connection_internal.h" ]
deps = [
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":jsep_transport_controller",
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
":peer_connection_message_handler",
":rtp_transceiver",
":rtp_transmission_manager",
":sctp_data_channel",
"../api:libjingle_peerconnection_api",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
"../call:call_interfaces",
Add RTCAudioPlayoutStats to GetStats(). This is done by allowing implementations of AudioDeviceModule to implement the GetStats() method. The default implementation returns nullopt, in which case RTCAudioPlayoutStats will not be visible in the stats. Bug: webrtc:14653 Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-13 16:42:36 +01:00
"../modules/audio_device",
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 10:32:30 +00:00
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
rtc_source_set("rtc_stats_collector") {
Remove internal dependencies on pc:peerconnection target This CL replaces those references with the smallest set of targets that can satisfy the linker dependencies revealed by building the "all" target. Bug: webrtc:13634 Change-Id: Ia778630b18e1164138c41d245c3c8effed67f8e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257282 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36445}
2022-03-31 19:53:50 +00:00
visibility = [
":*",
"../api:*",
]
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
sources = [
"rtc_stats_collector.cc",
"rtc_stats_collector.h",
]
deps = [
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
":data_channel_utils",
":peer_connection_internal",
":rtc_stats_traversal",
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
":rtp_transceiver",
":sctp_data_channel",
":track_media_info_map",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":transport_stats",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
":webrtc_sdp",
"../api:array_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:dtls_transport_interface",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing_statistics",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../api/task_queue:task_queue",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/units:time_delta",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../api/video:video_rtp_headers",
Add scalability mode to RTCOutboundRtpStreamStats stats This is in the webrtc-stats spec at https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode. This adds the scalability mode to CodecSpecificInfo which is used to plumb the modes for each simulcast layer. TBR=orphis@webrtc.org Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals. Bug: webrtc:14730 Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-06 10:09:10 +00:00
"../api/video_codecs:scalability_mode",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../call:call_interfaces",
"../common_video:common_video",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
"../media:media_channel_impl",
Add RTCAudioPlayoutStats to GetStats(). This is done by allowing implementations of AudioDeviceModule to implement the GetStats() method. The default implementation returns nullopt, in which case RTCAudioPlayoutStats will not be visible in the stats. Bug: webrtc:14653 Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-13 16:42:36 +01:00
"../modules/audio_device",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../rtc_base:checks",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:network_constants",
"../rtc_base:refcount",
"../rtc_base:rtc_event",
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
Use void* instead of uintptr_t for tracking pointers. RTCStatsCollector internally keeps track of open data channels but does not need (or want) to interact directly with those channels, hence uintptr_t was used instead of pointers to the channel objects. This changes that to use void* to avoid having to do the cast. This is a follow-up action item to https://webrtc-review.googlesource.com/c/src/+/295781 This CL also changes the container type: std::set -> webrtc::flat_set Bug: webrtc:12689 Change-Id: I13d3f4a41ef83dab38411193187e872b9d6d3cff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295871 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39468}
2023-03-03 13:34:42 +01:00
"../rtc_base/containers:flat_set",
Fix race in RTCStatsCollector's cache. `cached_certificates_by_transport_` is used on the network thread, but can be cleared from the signaling thread. To fix the race where clear happens at the same time as stats collecting, a mutex is added. This mutex should very rarely be contended in practise since ClearCachedStatsReport() typically only happen during renegotiation (e.g. when someone joins/leaves) and getStats only happens once per second or less (typically). NOTRY=Everything passes except unrelated purple bot Bug: webrtc:14510 Change-Id: Iaf539a5cc8c87184fa0a87b9c889a13b941a9ad1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38262}
2022-09-30 17:22:44 +02:00
"../rtc_base/synchronization:mutex",
Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I9043aa507421a93f0d7ba7406e237f727999b696 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268121 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37478}
2022-07-07 10:08:49 +02:00
"//third_party/abseil-cpp/absl/functional:bind_front",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
]
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
}
rtc_source_set("rtc_stats_traversal") {
visibility = [ ":*" ]
sources = [
"rtc_stats_traversal.cc",
"rtc_stats_traversal.h",
]
deps = [
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../rtc_base:checks",
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
rtc_source_set("sdp_offer_answer") {
visibility = [ ":*" ]
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
sources = [
"sdp_offer_answer.cc", # TODO: Make separate target when not circular
"sdp_offer_answer.h", # dependent on peerconnection.h
]
deps = [
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":connection_context",
":data_channel_controller",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":dtls_transport",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":jsep_transport_controller",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":media_stream",
":media_stream_observer",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":media_stream_proxy",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":peer_connection_internal",
":peer_connection_message_handler",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":rtp_media_utils",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":rtp_transceiver",
":rtp_transmission_manager",
":sdp_state_provider",
":session_description",
":simulcast_description",
":stream_collection",
":transceiver_list",
":usage_pattern",
Unify RTP payload type validity checking making the UsedId generator the source of truth. BUG=webrtc:12197 Change-Id: I4318a1366f8b2e20ea5ae264232437a9006c5103 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321120 Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40802}
2023-09-25 14:42:51 +02:00
":used_ids",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
":webrtc_session_description_factory",
"../api:array_view",
"../api:audio_options_api",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:dtls_transport_interface",
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 10:32:30 +00:00
"../api:field_trials_view",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../api:libjingle_peerconnection_api",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:make_ref_counted",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocator_factory",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api/video:video_codec_constants",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../call:payload_type",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../media:media_engine",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rid_description",
"../media:stream_params",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_description_factory",
"../p2p:transport_info",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_operations_chain",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../rtc_base:stringutils",
"../rtc_base:threading",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/algorithm:container",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
Break out sdp_offer_answer from peerconnection Bug: webrtc:13634 Change-Id: I4400c4d1a6aaf751774dbd8a9f697fd9fee1d297 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251324 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35954}
2022-02-08 12:55:42 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("jsep_ice_candidate") {
visibility = [ ":*" ]
}
rtc_source_set("jsep_session_description") {
visibility = [ ":*" ]
}
rtc_source_set("local_audio_source") {
visibility = [ ":*" ]
Break out local_audio_source from peerconnection target Bug: webrtc:13634 Change-Id: I402e5334869c69b8e7436f3402b04a93f8df7135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250201 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35887}
2022-02-02 11:26:03 +00:00
sources = [
"local_audio_source.cc",
"local_audio_source.h",
]
deps = [
"../api:audio_options_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("peer_connection") {
visibility = [ ":*" ]
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
sources = [
"peer_connection.cc",
"peer_connection.h",
]
deps = [
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":connection_context",
":data_channel_controller",
":data_channel_utils",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":dtls_transport",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":ice_server_parsing",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":jsep_transport_controller",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":peer_connection_internal",
":peer_connection_message_handler",
":rtc_stats_collector",
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":rtp_transceiver",
":rtp_transmission_manager",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":rtp_transport_internal",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":sctp_data_channel",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
":sctp_transport",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":sdp_offer_answer",
":session_description",
":simulcast_description",
":transceiver_list",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":transport_stats",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":usage_pattern",
":webrtc_session_description_factory",
"../api:async_dns_resolver",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:dtls_transport_interface",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:make_ref_counted",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:turn_customizer",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api/adaptation:resource_adaptation_api",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/audio:audio_device",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api/crypto:options",
Create PeerConnection specific environment Bug: webrtc:15656 Change-Id: I11616e3470798b43cb07a776f5d58669d629e24d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328960 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41283}
2023-11-30 08:59:39 +01:00
"../api/environment",
Remove some dependencies that aren't needed This was a side effect of testing out the "gn_check_autofix.py" tool after running "apply-iwyu -r" on a few files. Seems worth committing. Bug: none Change-Id: I3df446c640d4c4e3d6b15eddbdf66a1a40135f69 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258024 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36446}
2022-04-05 13:00:54 +00:00
"../api/rtc_event_log",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/transport:bandwidth_estimation_settings",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api/transport:bitrate_settings",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"../api/transport:network_control",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/units:time_delta",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api/video:video_codec_constants",
"../call:call_interfaces",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../call:payload_type_picker",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../media:codec",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rid_description",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../media:rtc_media_config",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:stream_params",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:basic_async_resolver_factory",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:connection_info",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:net_helper",
Ensure <sys/socket.h> is included by using "rtc_base/net_helpers.h". * IWYU export <sys/socket.h> from rtc_base/net_helpers.h. * Add a presubmit check to ensures that <sys/socket.h> is included through net_helpers.h (expect if there is a IWYU pragma or a no-presubmit-check). * Clean up existing includes of <sys/socket.h> Change-Id: I4bc6cce045c046287f8f74f89edfc9321293b274 Bug: b/236227627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362082 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#42996}
2024-09-10 09:47:49 +02:00
"../rtc_base:net_helpers",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:network_constants",
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:stringutils",
"../rtc_base:threading",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/algorithm:container",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
Rename simulcast SDP serializer which is not a generic SDP serializer but only deals with the simulcast SDP. BUG=None Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40761}
2023-09-14 17:23:59 +02:00
rtc_source_set("simulcast_sdp_serializer") {
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
visibility = [ ":*" ]
Break out sdp_serializer target Bug: webrtc:13634 Change-Id: I6bf28cb8b37257800c7944b870f9d2cb83273dfb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35880}
2022-02-01 20:59:51 +00:00
sources = [
Rename simulcast SDP serializer which is not a generic SDP serializer but only deals with the simulcast SDP. BUG=None Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40761}
2023-09-14 17:23:59 +02:00
"simulcast_sdp_serializer.cc",
"simulcast_sdp_serializer.h",
Break out sdp_serializer target Bug: webrtc:13634 Change-Id: I6bf28cb8b37257800c7944b870f9d2cb83273dfb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35880}
2022-02-01 20:59:51 +00:00
]
deps = [
":session_description",
":simulcast_description",
"../api:rtc_error",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rid_description",
Break out sdp_serializer target Bug: webrtc:13634 Change-Id: I6bf28cb8b37257800c7944b870f9d2cb83273dfb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35880}
2022-02-01 20:59:51 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/algorithm:container",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out sdp_serializer target Bug: webrtc:13634 Change-Id: I6bf28cb8b37257800c7944b870f9d2cb83273dfb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250161 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35880}
2022-02-01 20:59:51 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("sdp_utils") {
Break out sdp_utils from peerconnection target Bug: webrtc:13634 Change-Id: Ibf337e93ac0581ebaf83b3f5f978704ae73ce9dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35896}
2022-02-03 07:56:22 +00:00
sources = [
"sdp_utils.cc",
"sdp_utils.h",
]
deps = [
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
":session_description",
Break out sdp_utils from peerconnection target Bug: webrtc:13634 Change-Id: Ibf337e93ac0581ebaf83b3f5f978704ae73ce9dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35896}
2022-02-03 07:56:22 +00:00
"../api:libjingle_peerconnection_api",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Break out sdp_utils from peerconnection target Bug: webrtc:13634 Change-Id: Ibf337e93ac0581ebaf83b3f5f978704ae73ce9dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35896}
2022-02-03 07:56:22 +00:00
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
rtc_source_set("legacy_stats_collector") {
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
visibility = [ ":*" ]
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
sources = [
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
"legacy_stats_collector.cc",
"legacy_stats_collector.h",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
]
deps = [
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
":data_channel_utils",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector_interface",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
":peer_connection_internal",
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
":rtp_sender_proxy",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
":rtp_transceiver",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":transport_stats",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../api:scoped_refptr",
"../api:sequence_checker",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing_statistics",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../api/audio_codecs:audio_codecs_api",
"../api/video:video_rtp_headers",
"../call:call_interfaces",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../rtc_base:checks",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../rtc_base:ip_address",
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../rtc_base:network_constants",
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out stats_collector from pc/peerconnection target Bug: webrtc:13634 Change-Id: I868ea6e911b81da3a9ecc39ae9880f5d4d718302 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35948}
2022-02-08 09:30:41 +00:00
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("stream_collection") {
visibility = [ ":*" ]
Break out stream_collection.h from peerconnection target Bug: webrtc:13634 Change-Id: I7062f245b3ae5f1d7f38baee0c5b14be3332e658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251061 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35925}
2022-02-07 06:43:48 +00:00
sources = [ "stream_collection.h" ]
deps = [ "../api:libjingle_peerconnection_api" ]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("track_media_info_map") {
visibility = [ ":*" ]
Break out track_media_info_map from peerconnection Bug: webrtc:13634 Change-Id: If0a24738fbbff0b3706c3926e7188b8b6a451cdc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35904}
2022-02-03 19:02:38 +00:00
sources = [
"track_media_info_map.cc",
"track_media_info_map.h",
]
deps = [
":rtp_receiver",
":rtp_sender",
TrackMediaInfoMap: Use rtc::ArrayView in Initialize. Drive-by improvement as suggested in https://webrtc-review.googlesource.com/c/src/+/269404. Bug: webrtc:14289 Change-Id: Ib6579916cb4ab1076c1522275b318859400b731e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269202 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37625}
2022-07-27 11:32:14 +02:00
"../api:array_view",
Break out track_media_info_map from peerconnection Bug: webrtc:13634 Change-Id: If0a24738fbbff0b3706c3926e7188b8b6a451cdc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35904}
2022-02-03 19:02:38 +00:00
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
"../media:stream_params",
Break out track_media_info_map from peerconnection Bug: webrtc:13634 Change-Id: If0a24738fbbff0b3706c3926e7188b8b6a451cdc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35904}
2022-02-03 19:02:38 +00:00
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:threading",
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("webrtc_sdp") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Mark webrtc_sdp as visible target It is being used, among others, by Chrome. Bug: webrtc:13634 Change-Id: I61954e3a149ead7abf47efe67d6fbecd63ad8d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35928}
2022-02-07 08:39:52 +00:00
visibility = [ "*" ] # Used by Chrome and more
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
sources = [
"jsep_ice_candidate.cc",
"jsep_session_description.cc",
"webrtc_sdp.cc",
"webrtc_sdp.h",
]
deps = [
":media_protocol_names",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
":session_description",
":simulcast_description",
Rename simulcast SDP serializer which is not a generic SDP serializer but only deals with the simulcast SDP. BUG=None Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40761}
2023-09-14 17:23:59 +02:00
":simulcast_sdp_serializer",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_constants",
"../media:rid_description",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../media:rtc_data_sctp_transport_internal",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rtp_utils",
"../media:stream_params",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:candidate_pair_interface",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:net_helper",
Add generation of CCFB in answer (RFC8888 support) Note: This still doesn't enable CCFB - it just completes the signalling. Bug: webrtc:42225697 Change-Id: I2dfd346075f2adcc438588f592c8f735f4101c89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367260 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43348}
2024-11-01 13:13:57 +00:00
"../rtc_base:net_helpers",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../rtc_base:network_constants",
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/algorithm:container",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"//third_party/abseil-cpp/absl/strings",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_source_set("webrtc_session_description_factory") {
visibility = [ ":*" ]
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
sources = [
"webrtc_session_description_factory.cc",
"webrtc_session_description_factory.h",
]
deps = [
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
":connection_context",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
":sdp_state_provider",
":session_description",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../api:field_trials_view",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:scoped_refptr",
Remove //api:sequence_checker from public deps Bug: webrtc:8603 Change-Id: Ib572cdc5b5f4c1f0eb3357e5e0812fc9ce7437c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257915 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36465}
2022-04-04 17:18:21 +02:00
"../api:sequence_checker",
Remove dependency on rtc::MessageHandler in session description factory Bug: webrtc:9702 Change-Id: Iedbcc1f8d223c4df3e0e8c5811d5a4b78dfe8d3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272620 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37891}
2022-08-22 19:45:43 +02:00
"../api/task_queue",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../call:payload_type",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_description_factory",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:rtc_certificate_generator",
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
"../rtc_base:stringutils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Remove a sigslot from webrtc_session_description_factory callback are know at construction time and only need some synchronization at destruction time. In this case such synchronization can be done with cheaper/simpler WeakPtr concept. Asynchronous call to SetCertificate is no longer needed thanks to previous removal of sigslot in https://webrtc-review.googlesource.com/c/src/+/192362 Bug: webrtc:11943 Change-Id: Icadbcb4f83be9ed4b8f53a72beaef8573f2c9356 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272402 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37868}
2022-08-19 14:03:54 +02:00
"../rtc_base:weak_ptr",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Remove dependency on rtc::MessageHandler in session description factory Bug: webrtc:9702 Change-Id: Iedbcc1f8d223c4df3e0e8c5811d5a4b78dfe8d3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272620 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37891}
2022-08-22 19:45:43 +02:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
Break out webrtc_session-description_factory from peerconnection target Bug: webrtc:13634 Change-Id: I8d13c0c86f1883aeb40a6055c8f406aa38d34338 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250441 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35899}
2022-02-03 09:48:25 +00:00
]
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
}
rtc_library("ice_server_parsing") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
visibility = [ "*" ] # Known to be used externally
Mark webrtc_sdp as visible target It is being used, among others, by Chrome. Bug: webrtc:13634 Change-Id: I61954e3a149ead7abf47efe67d6fbecd63ad8d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35928}
2022-02-07 08:39:52 +00:00
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
sources = [
"ice_server_parsing.cc",
"ice_server_parsing.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
"../rtc_base:checks",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:socket_address",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
]
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
}
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
rtc_library("media_stream_observer") {
Break out media_stream_observer.cc in peerconnection build Bug: webrtc:13634 Change-Id: Ie1368217bc3735b0a747c39dbbb5afbc929c4cf3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250080 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35867}
2022-02-01 07:47:53 +00:00
sources = [
"media_stream_observer.cc",
"media_stream_observer.h",
]
deps = [
"../api:media_stream_interface",
"../api:scoped_refptr",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Break out media_stream_observer.cc in peerconnection build Bug: webrtc:13634 Change-Id: Ie1368217bc3735b0a747c39dbbb5afbc929c4cf3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250080 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35867}
2022-02-01 07:47:53 +00:00
]
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
}
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
rtc_source_set("peer_connection_factory") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
visibility = [ "*" ] # Known to be used externally
Add environment_construction poison This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation. This poison extends and thus replaces default task queue poison. Bug: webrtc:15656 Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41247}
2023-11-27 10:57:22 +01:00
allow_poison = [ "environment_construction" ]
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
sources = [
"peer_connection_factory.cc",
"peer_connection_factory.h",
]
deps = [
":local_audio_source",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":media_stream_proxy",
":media_stream_track_proxy",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":peer_connection",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":peer_connection_factory_proxy",
":peer_connection_proxy",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api:audio_options_api",
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:network_state_predictor_api",
"../api:packet_socket_factory",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
Use Environment to keep peer connection factory field trials in ConnectionContext Bug: webrtc:15656 Change-Id: Ice52fcb9ba54a5d0034b59233ceae4f9cefbceae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328860 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41252}
2023-11-27 15:00:21 +01:00
"../api/environment",
Add environment_construction poison This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation. This poison extends and thus replaces default task queue poison. Bug: webrtc:15656 Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41247}
2023-11-27 10:57:22 +01:00
"../api/environment:environment_factory",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/metronome",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../api/neteq:neteq_api",
"../api/rtc_event_log:rtc_event_log",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/transport:sctp_transport_factory_interface",
"../api/units:data_rate",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../call:rtp_sender",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:basic_packet_socket_factory",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:basic_port_allocator",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:default_ice_transport_factory",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port_allocator",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../pc:audio_track",
"../pc:connection_context",
Delete deprecated call_factory and media_engine from PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: Id0ead8086ddd41f6792e2a3c224d8705cd797d49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326003 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41309}
2023-12-04 12:05:38 +01:00
"../pc:media_factory",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../pc:media_stream",
"../pc:rtp_parameters_conversion",
"../pc:session_description",
"../pc:video_track",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:rtc_certificate_generator",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
"../rtc_base:safe_conversions",
"../rtc_base:threading",
"../rtc_base/experiments:field_trial_parser",
"../rtc_base/system:file_wrapper",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
]
}
Add dummy targets for candidate breakouts Bug: webrtc:13634 Change-Id: I954e1f5b953805b55a5931a806ac71d9275be30a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249947 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35852}
2022-01-31 10:58:51 +00:00
Separate PeerConnection's self-message functions to a new file This prevents having to have sdp_offer_answer depend on peer_connection for the messaging functions. Bug: webrtc:11995 Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32310}
2020-10-05 07:01:09 +00:00
rtc_library("peer_connection_message_handler") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Separate PeerConnection's self-message functions to a new file This prevents having to have sdp_offer_answer depend on peer_connection for the messaging functions. Bug: webrtc:11995 Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32310}
2020-10-05 07:01:09 +00:00
sources = [
"peer_connection_message_handler.cc",
"peer_connection_message_handler.h",
]
deps = [
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector_interface",
Separate PeerConnection's self-message functions to a new file This prevents having to have sdp_offer_answer depend on peer_connection for the messaging functions. Bug: webrtc:11995 Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32310}
2020-10-05 07:01:09 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Rewrite PeerConnectionMessageHandler to not use rtc::MessageHandler Bug: webrtc:9702 Change-Id: I92390262b4794b1061702663621a9a4db22d367f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272023 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37836}
2022-08-17 14:58:40 +02:00
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:checks",
Separate PeerConnection's self-message functions to a new file This prevents having to have sdp_offer_answer depend on peer_connection for the messaging functions. Bug: webrtc:11995 Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32310}
2020-10-05 07:01:09 +00:00
]
}
Move the PeerConnection's usage pattern concept to its own file. This makes it easier to use it from multiple other modules. Bug: webrtc:11995 Change-Id: Id23843ae4600ebe46aed7465e873d107089fd50b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187347 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32361}
2020-10-09 05:43:53 +00:00
rtc_library("usage_pattern") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Move the PeerConnection's usage pattern concept to its own file. This makes it easier to use it from multiple other modules. Bug: webrtc:11995 Change-Id: Id23843ae4600ebe46aed7465e873d107089fd50b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187347 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32361}
2020-10-09 05:43:53 +00:00
sources = [
"usage_pattern.cc",
"usage_pattern.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../rtc_base:logging",
"../system_wrappers:metrics",
]
}
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
rtc_library("rtp_transceiver") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
sources = [
"rtp_transceiver.cc",
"rtp_transceiver.h",
]
deps = [
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 10:32:30 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Remove remaining trampoline functions from channel_manager This is part of the project to delete the class entirely. The CL also adds an "use_rtx" parameter to the function for listing video codecs, rather than filtering those away afterwards. Bug: webrtc:13931 Change-Id: I96b9b18c694a1c0986ccf22face76ef4c704d372 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262666 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36963}
2022-05-22 10:57:01 +00:00
":connection_context",
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
":proxy",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":rtp_media_utils",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
":rtp_parameters_conversion",
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":rtp_transport_internal",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
":session_description",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:array_view",
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 10:32:30 +00:00
"../api:audio_options_api",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:field_trials_view",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"../api:libjingle_peerconnection_api",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:rtc_error",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:options",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/task_queue",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757}
2022-05-04 10:32:30 +00:00
"../api/video:video_bitrate_allocator_factory",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api/video_codecs:scalability_mode",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../media:media_channel_impl",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../media:rtc_media_config",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Add dependencies on absl when they are used but undeclared Bug: b/36882554 Change-Id: I3a1c5f0024abc452bcd74eef2b66d4493f4f974c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37320}
2022-06-24 00:51:28 +02:00
"//third_party/abseil-cpp/absl/memory",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
]
Add dummy target that clients can depend on This will turn into a real target later. Bug: webrtc:11967 Change-Id: I518e2ab5f2cbb19245683d9e48f2e2555fbe99d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187342 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32351}
2020-10-08 12:01:06 +00:00
}
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
rtc_library("rtp_transmission_manager") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
sources = [
"rtp_transmission_manager.cc",
"rtp_transmission_manager.h",
]
deps = [
":audio_rtp_receiver",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Eliminate channel.h from rtp_transmission_manager.cc This also hides the existence of the classes VideoChannel and VoiceChannel from anything that does not include "channel.h". Bug: webrtc:13931 Change-Id: I080a692b6acfd5d2d0401ec20d59c3a684eddb05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260944 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36746}
2022-05-03 13:44:34 +00:00
":channel_interface",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector_interface",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
":rtp_receiver",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_receiver_proxy",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
":rtp_transceiver",
":transceiver_list",
":usage_pattern",
":video_rtp_receiver",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Propagate environment to RtpSenders Will be later used to conditionally enable mixed codec simulcast with a field trial. Bug: webrtc:42220378 Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42929}
2024-09-03 09:13:23 +00:00
"../api/environment",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:weak_ptr",
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
]
}
rtc_library("transceiver_list") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
sources = [
"transceiver_list.cc",
"transceiver_list.h",
]
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
deps = [
":rtp_transceiver",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:scoped_refptr",
Add threading assertions to TransceiverList Also add a function for accessing the list as internal transceivers rather than accessing the proxy objects; this exposes where the internal objects are accessed and where we need external references. Used the new list function in sdp_offer_answer wherever possible. Adds an UnsafeList function that is not thread guarded, so that the job of rooting out those instances can be done in a later CL. Bug: webrtc:12692 Change-Id: Ia591f22a1c8f82ec452a1a66a94fbf9ab9debd14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215581 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33781}
2021-04-19 21:21:36 +00:00
"../api:sequence_checker",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:checks",
Add threading assertions to TransceiverList Also add a function for accessing the list as internal transceivers rather than accessing the proxy objects; this exposes where the internal objects are accessed and where we need external references. Used the new list function in sdp_offer_answer wherever possible. Adds an UnsafeList function that is not thread guarded, so that the job of rooting out those instances can be done in a later CL. Bug: webrtc:12692 Change-Id: Ia591f22a1c8f82ec452a1a66a94fbf9ab9debd14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215581 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33781}
2021-04-19 21:21:36 +00:00
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
]
Separate RTP object handling (senders, receivers, transceivers) This is part of the PeerConnection disassembly project. Bug: webrtc:11995 Change-Id: I4f207c8af39e267c4b5752c0828b84e221e1f080 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188624 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32443}
2020-10-19 13:28:05 +00:00
}
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
rtc_library("rtp_receiver") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
sources = [
"rtp_receiver.cc",
"rtp_receiver.h",
]
deps = [
":media_stream",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":media_stream_proxy",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
":video_track_source",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/video:video_frame",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
"../media:video_broadcaster",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base:checks",
"../rtc_base:logging",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
]
}
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
rtc_library("audio_rtp_receiver") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"audio_rtp_receiver.cc",
"audio_rtp_receiver.h",
]
deps = [
":audio_track",
":jitter_buffer_delay",
":media_stream",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":media_stream_track_proxy",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":remote_audio_source",
":rtp_receiver",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:frame_transformer_interface",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api/crypto:frame_decryptor_interface",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/transport/rtp:rtp_source",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
rtc_library("video_rtp_receiver") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"video_rtp_receiver.cc",
"video_rtp_receiver.h",
]
deps = [
":jitter_buffer_delay",
":media_stream",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":media_stream_track_proxy",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":rtp_receiver",
":video_rtp_track_source",
":video_track",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api/crypto:frame_decryptor_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/transport/rtp:rtp_source",
"../api/video:recordable_encoded_frame",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api/video:video_frame",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
rtc_library("video_rtp_track_source") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"video_rtp_track_source.cc",
"video_rtp_track_source.h",
]
deps = [
":video_track_source",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
"../media:video_broadcaster",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:checks",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base/synchronization:mutex",
Reland "Introduce RTC_NO_UNIQUE_ADDRESS." This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447 This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since there have been some issues. Original change's description: > Introduce RTC_NO_UNIQUE_ADDRESS. > > This macro introduces the possibility to suggest the compiler that a > data member doesn't need an address different from other non static > data members. > > The usage of a macro is to maintain portability since at the moment > the attribute [[no_unique_address]] is only supported by clang > with at least -std=c++11 but it should be supported by all the > compilers starting from C++20. > > Bug: webrtc:11495 > Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32246} Bug: webrtc:11495, webrtc:12218 Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:07:42 +01:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
rtc_library("audio_track") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"audio_track.cc",
"audio_track.h",
]
deps = [
"../api:media_stream_interface",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Add several thread checks to RtpSender classes. Minor related updates to AudioTrack and VideoTrack's sequence checkers. There's more that can be done (or arguably needs to), but this is a start. Bug: none Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36021}
2022-02-16 20:02:12 +01:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
rtc_library("video_track") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"video_track.cc",
"video_track.h",
]
deps = [
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":video_track_source_proxy",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api:media_stream_interface",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api/video:video_frame",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:video_source_base",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Add several thread checks to RtpSender classes. Minor related updates to AudioTrack and VideoTrack's sequence checkers. There's more that can be done (or arguably needs to), but this is a start. Bug: none Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36021}
2022-02-16 20:02:12 +01:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
Break circular dependency on WebRtcSessionDescriptionFactory After this change, SdpOfferAnswerHandler implements a read-only interface called SdpStateProvider, which allows enough access for WebRtcSessionDescriptionFactory to learn what it needs to know. Bug: webrtc:12060 Change-Id: Ic888b5027b2df5fee407d32b89da66ff044c40de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190145 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32486}
2020-10-23 13:30:46 +00:00
rtc_source_set("sdp_state_provider") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break circular dependency on WebRtcSessionDescriptionFactory After this change, SdpOfferAnswerHandler implements a read-only interface called SdpStateProvider, which allows enough access for WebRtcSessionDescriptionFactory to learn what it needs to know. Bug: webrtc:12060 Change-Id: Ic888b5027b2df5fee407d32b89da66ff044c40de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190145 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32486}
2020-10-23 13:30:46 +00:00
sources = [ "sdp_state_provider.h" ]
Remove the rtc_pc_base target This target is no longer needed. Bug: webrtc:13805 Change-Id: I6247fbdaf630bad88510e2527d5a35330260896d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257140 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36389}
2022-03-30 07:33:52 +00:00
deps = [ "../api:libjingle_peerconnection_api" ]
Break circular dependency on WebRtcSessionDescriptionFactory After this change, SdpOfferAnswerHandler implements a read-only interface called SdpStateProvider, which allows enough access for WebRtcSessionDescriptionFactory to learn what it needs to know. Bug: webrtc:12060 Change-Id: Ic888b5027b2df5fee407d32b89da66ff044c40de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190145 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32486}
2020-10-23 13:30:46 +00:00
}
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
rtc_library("jitter_buffer_delay") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"jitter_buffer_delay.cc",
"jitter_buffer_delay.h",
]
deps = [
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:macromagic",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base:safe_conversions",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:safe_minmax",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
]
}
rtc_library("remote_audio_source") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
sources = [
"remote_audio_source.cc",
"remote_audio_source.h",
]
deps = [
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../api:call_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Remove usage of rtc::MessageHandler in pc/remote_audio_source Bug: webrtc:9702 Change-Id: Ibef43b8c1b61afe4cf4e79a7c6549af6d5bff93f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272546 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37859}
2022-08-22 10:22:40 +02:00
"../api/task_queue",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:checks",
Audio stack traces Bug: webrtc:0 Change-Id: I90ea6301f02c2ebe72711ddbeda0bf000a6873aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276940 Auto-Submit: Olga Sharonova <olka@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38223}
2022-09-27 15:22:34 +02:00
"../rtc_base:event_tracer",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
"../rtc_base:logging",
"../rtc_base:safe_conversions",
"../rtc_base:stringutils",
"../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
rtc_library("rtp_sender") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
sources = [
"rtp_sender.cc",
"rtp_sender.h",
]
deps = [
":dtmf_sender",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector_interface",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../api:audio_options_api",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:dtmf_sender_interface",
Propagate environment to RtpSenders Will be later used to conditionally enable mixed codec simulcast with a field trial. Bug: webrtc:42220378 Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42929}
2024-09-03 09:13:23 +00:00
"../api:field_trials_view",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:frame_transformer_interface",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:priority",
"../api:rtc_error",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:scoped_refptr",
Remove //api:sequence_checker from public deps Bug: webrtc:8603 Change-Id: Ib572cdc5b5f4c1f0eb3357e5e0812fc9ce7437c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257915 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36465}
2022-04-04 17:18:21 +02:00
"../api:sequence_checker",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/crypto:frame_encryptor_interface",
Propagate environment to RtpSenders Will be later used to conditionally enable mixed codec simulcast with a field trial. Bug: webrtc:42220378 Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42929}
2024-09-03 09:13:23 +00:00
"../api/environment",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:audio_source",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base:threading",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base/synchronization:mutex",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base/third_party/sigslot",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
rtc_library("rtp_parameters_conversion") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
sources = [
"rtp_parameters_conversion.cc",
"rtp_parameters_conversion.h",
]
deps = [
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
":session_description",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_constants",
"../media:rtp_utils",
"../media:stream_params",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
]
}
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
rtc_library("dtmf_sender") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
sources = [
"dtmf_sender.cc",
"dtmf_sender.h",
]
deps = [
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
":proxy",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:dtmf_sender_interface",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../api:libjingle_peerconnection_api",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api:scoped_refptr",
Remove //api:sequence_checker from public deps Bug: webrtc:8603 Change-Id: Ib572cdc5b5f4c1f0eb3357e5e0812fc9ce7437c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257915 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36465}
2022-04-04 17:18:21 +02:00
"../api:sequence_checker",
Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I9043aa507421a93f0d7ba7406e237f727999b696 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268121 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37478}
2022-07-07 10:08:49 +02:00
"../api/task_queue",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I9043aa507421a93f0d7ba7406e237f727999b696 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268121 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37478}
2022-07-07 10:08:49 +02:00
"../api/units:time_delta",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../rtc_base:refcount",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../rtc_base/third_party/sigslot",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
]
}
rtc_library("media_stream") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
sources = [
"media_stream.cc",
"media_stream.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../rtc_base:checks",
]
}
rtc_library("video_track_source") {
sources = [
"video_track_source.cc",
"video_track_source.h",
]
deps = [
"../api:media_stream_interface",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../api:sequence_checker",
Running apply-iwyu on ~all files in pc/ Bug: none Change-Id: Ieebdfb743e691f7ae35e1aa354f68ce9e771064d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33105}
2021-01-29 14:45:08 +00:00
"../api/video:recordable_encoded_frame",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../api/video:video_frame",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base:checks",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." This is a reland of 3ed36c0521546881656c73984456485dcab16205 Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f > > The difference from the original CL is that a check for > `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed. > This caused a side effect that registering the sink while the source > was in an "initializing" state, failed. The last remaining state > however, is `kEnded` - but since there's no logic in the class around > the expected value of the states, the check inside of AddSink() > doesn't provide an additional value - it's rather a surprise for > developers if it doesn't succeed. So, now removed. > > Original change's description: > > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver. > > > > This simplifies the logic in these classes a bit, which makes upcoming > > change easier. The `stopped_` flag in these classes was essentially > > the same thing as `media_channel_ == nullptr`, which is what's > > consistently used now for the same checks. > > > > Bug: webrtc:13540 > > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35907} > > Bug: webrtc:13540 > Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35958} Bug: webrtc:13540 Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35967}
2022-02-08 21:12:15 +01:00
"../rtc_base/system:no_unique_address",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
"../rtc_base/system:rtc_export",
]
}
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
rtc_source_set("legacy_stats_collector_interface") {
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ ":*" ]
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
sources = [ "legacy_stats_collector_interface.h" ]
Separate PeerConnection's self-message functions to a new file This prevents having to have sdp_offer_answer depend on peer_connection for the messaging functions. Bug: webrtc:11995 Change-Id: Icad7c9c0e6149bd1b8d78e37eff5f9786b74692e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186662 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32310}
2020-10-05 07:01:09 +00:00
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
]
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
}
Revise the pc:libjingle_peerconnection target This adds the absolutely required files for this target that is used by Chrome and others in order to link in all needed libraries, and removes the dependency on peerconnection. Bug: webrtc:13634 Change-Id: Ia66f5f627680ce15bcac941998ca1b6da4edb6ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331621 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41395}
2023-12-15 14:39:18 +00:00
# This target contains the libraries that are required in order to get an
# usable peerconnection-using binary.
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
rtc_source_set("libjingle_peerconnection") {
Mark webrtc_sdp as visible target It is being used, among others, by Chrome. Bug: webrtc:13634 Change-Id: I61954e3a149ead7abf47efe67d6fbecd63ad8d2a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35928}
2022-02-07 08:39:52 +00:00
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
visibility = [ "*" ] # Used by Chrome and others
Add environment_construction poison This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation. This poison extends and thus replaces default task queue poison. Bug: webrtc:15656 Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41247}
2023-11-27 10:57:22 +01:00
allow_poison = [ "environment_construction" ]
Add restrictive visibility to all targets in //pc This CL sets all visibility to ":*" (this buildfile) where no users outside this directory are known, and marks up publicly exported targets and Chrome dependencies explicitly. Bug: webrtc:13661 Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36004}
2022-02-15 12:45:02 +00:00
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
deps = [
Revise the pc:libjingle_peerconnection target This adds the absolutely required files for this target that is used by Chrome and others in order to link in all needed libraries, and removes the dependency on peerconnection. Bug: webrtc:13634 Change-Id: Ia66f5f627680ce15bcac941998ca1b6da4edb6ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331621 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41395}
2023-12-15 14:39:18 +00:00
":jsep_session_description",
":peer_connection_factory",
":rtc_stats_collector",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
"../api:libjingle_peerconnection_api",
Revise the pc:libjingle_peerconnection target This adds the absolutely required files for this target that is used by Chrome and others in order to link in all needed libraries, and removes the dependency on peerconnection. Bug: webrtc:13634 Change-Id: Ia66f5f627680ce15bcac941998ca1b6da4edb6ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331621 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41395}
2023-12-15 14:39:18 +00:00
"../stats",
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
]
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
}
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
if (rtc_include_tests && !build_with_chromium) {
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("rtc_pc_unittests") {
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
testonly = true
sources = [
Add unit test of AudioRtpReceiver Bug: chromium:1272566 Change-Id: Ic10c586062cc8fdded2dd6249b49fab527c99823 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238987 Commit-Queue: Tony Herre <toprice@chromium.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35429}
2021-11-29 11:33:42 +01:00
"audio_rtp_receiver_unittest.cc",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
"channel_unittest.cc",
add DTLSSrtpTransport/SrtpTransport integration test which shows that a DtlsSrtpTransport can send and receive from the SrtpTransport which extracts the key from its DTLS transport. The SrtpTransport takes its keys from the DtlsSrtpTransport which (by the way of encryption and decryption) ensures both sides agree on the keys to use BUG=webrtc:357776213 Change-Id: I605c6ae660eab5a53bef69bcf84d7e70a34d7be1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365274 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43231}
2024-10-11 13:24:17 -07:00
"dtls_srtp_transport_integrationtest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"dtls_srtp_transport_unittest.cc",
Added underscore to dtls_transport_unittest.cc. Bug: webrtc:9860 Change-Id: Ic58361ae8daf44666af51708e6fd3f586f25c6df Reviewed-on: https://webrtc-review.googlesource.com/c/124625 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26880}
2019-02-26 18:20:05 -08:00
"dtls_transport_unittest.cc",
Add IceTransportInterface object This creates the API for an ICE transport object, and lets it be accessible from a DTLS transport object. Bug: chromium:907849 Change-Id: Ieb24570217dec75ce0deca8420739c1f116fbba4 Reviewed-on: https://webrtc-review.googlesource.com/c/118703 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26472}
2019-01-30 14:57:03 +01:00
"ice_transport_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"jsep_transport_controller_unittest.cc",
"jsep_transport_unittest.cc",
"media_session_unittest.cc",
"rtcp_mux_filter_unittest.cc",
"rtp_transport_unittest.cc",
Add SCTP transport to the public API. This involves inserting an extra layer between jsep_transport_controller and the cricket::SctpTransportInternal layer. The objects at this layer are reference counted. Bug: chromium:818643 Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39 Reviewed-on: https://webrtc-review.googlesource.com/c/123884 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26889}
2019-02-28 07:51:00 +01:00
"sctp_transport_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"session_description_unittest.cc",
"srtp_session_unittest.cc",
"srtp_transport_unittest.cc",
"test/rtp_transport_test_util.h",
"test/srtp_test_util.h",
Add support for two-byte RTP header extensions to UsedRtpHeaderExtensionIds Bug: webrtc:9985 Change-Id: I4c736cb920d1ac03e43c98a81218f037feebcd97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139601 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28346}
2019-06-20 15:37:52 +02:00
"used_ids_unittest.cc",
VideoRtpReceiver: Enable encoded frame sink. This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames can flow to sinks installed in VideoTrackSourceInterface. Bug: chromium:1013590 Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30003}
2019-12-04 12:57:58 +01:00
"video_rtp_receiver_unittest.cc",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
]
if (is_win) {
libs = [ "strmiids.lib" ]
}
deps = [
Add unit test of AudioRtpReceiver Bug: chromium:1272566 Change-Id: Ic10c586062cc8fdded2dd6249b49fab527c99823 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238987 Commit-Queue: Tony Herre <toprice@chromium.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35429}
2021-11-29 11:33:42 +01:00
":audio_rtp_receiver",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
":dtls_srtp_transport",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":dtls_transport",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":ice_transport",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":jsep_transport",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":jsep_transport_controller",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
":libjingle_peerconnection",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
":media_protocol_names",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Move the TransportController from p2p/base to pc/. The TransportController was in p2p/base before and it cannot depend on pc/ or media/ level targets because of the circular dependency. To make the TransportController be responsible for creating and managing the RtpTransport related objects which are pc/ level targets, the TransportController is moved from p2p/base to pc/. The TransportController makes more sense in pc/ anyway, since its main responsibility is processing the "transport" parts of SDP which is PeerConnection-specific. This is also easier than moving RtpTransport related objects to p2p/base because those objects also depend on other media/ and pc/ level targets such as srtpfilter, cryptoparams etc. Bug: webrtc:7013 Change-Id: Ic48dd5c454046ff3c81331f4b459f96a3255f328 Reviewed-on: https://webrtc-review.googlesource.com/4560 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20049}
2017-09-29 10:51:43 -07:00
":pc_test_utils",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
":rtc_pc",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":rtcp_mux_filter",
":rtp_media_utils",
Makes sure that RED is not added twice to the list of codecs when it is used with Opus. Bug: webrtc:15606 Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#41028}
2023-10-27 12:25:57 +00:00
":rtp_parameters_conversion",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
":rtp_transport",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":rtp_transport_internal",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
":sctp_transport",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
":session_description",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
":simulcast_description",
Break out remaining orange targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I97f000888c63209a557fbb6a428b176c2a79b5da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36225}
2022-03-15 13:21:59 +00:00
":srtp_session",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
":srtp_transport",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
":transport_stats",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":used_ids",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":video_rtp_receiver",
Move array_view.h to webrtc/api/ We use ArrayView in our public API, so its header should be in webrtc/api/. BUG=none Review-Url: https://codereview.webrtc.org/3007763002 Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 05:43:17 -07:00
"../api:array_view",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:audio_options_api",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
"../api:dtls_transport_interface",
Add IceTransportInterface object This creates the API for an ICE transport object, and lets it be accessible from a DTLS transport object. Bug: chromium:907849 Change-Id: Ieb24570217dec75ce0deca8420739c1f116fbba4 Reviewed-on: https://webrtc-review.googlesource.com/c/118703 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26472}
2019-01-30 14:57:03 +01:00
"../api:ice_transport_factory",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../api:ice_transport_interface",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
"../api:libjingle_peerconnection_api",
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219}
2022-06-14 15:48:26 +02:00
"../api:make_ref_counted",
"../api:make_ref_counted",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Move rtc_error.{h,cc} to its own build target. Bug: webrtc:8733 Change-Id: Idd34d9a88ae62a01b9ea50719872f8188069211e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152320 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29148}
2019-09-10 10:51:23 +02:00
"../api:rtc_error",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../api:rtp_headers",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../api:rtp_transceiver_direction",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api:scoped_refptr",
Remove //api:sequence_checker from public deps Bug: webrtc:8603 Change-Id: Ib572cdc5b5f4c1f0eb3357e5e0812fc9ce7437c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257915 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36465}
2022-04-04 17:18:21 +02:00
"../api:sequence_checker",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../api/audio_codecs:audio_codecs_api",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../api/crypto:options",
"../api/environment:environment",
sctp: Pass webrtc::Environment to DcSctpTransport The DcSctpTransport will soon use field trials to conditionally enable some options. And overall, there is a migration project to start using the Environment and this CL is in that direction, also setting the boundary; The dcSCTP library should not depend on it. But the transport is allowed to. Bug: webrtc:14997 Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41872}
2024-02-28 12:16:15 +01:00
"../api/environment:environment_factory",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/task_queue:task_queue",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
Make VideoBitrateAllocatorFactory injectable. This patch makes VideoBitrateAllocatorFactory injectable by adding to PeerConnectionDependencies instead of allowing it to be overridden using MediaEngine (on PeerConnectionFactory). With this patch VideoBitrateAllocatorFactory is owned by the PeerConnection. WANT_LGTM (examples) : sakal@ WANT_LGTM (api/pc) : steveanton@ Bug: webrtc:10547 Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27654}
2019-04-17 07:38:40 +02:00
"../api/video:builtin_video_bitrate_allocator_factory",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/video:recordable_encoded_frame",
VideoRtpReceiver: Enable encoded frame sink. This change finally wires up VideoRtpReceiver::OnGenerateKeyFrame and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames can flow to sinks installed in VideoTrackSourceInterface. Bug: chromium:1013590 Change-Id: I76f8226752294aee8fe137d1a78ee66548900cc2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161095 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30003}
2019-12-04 12:57:58 +01:00
"../api/video/test:mock_recordable_encoded_frame",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../call:payload_type_picker",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../call:rtp_interfaces",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../call:rtp_receiver",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../media:rtc_data_sctp_transport_internal",
Enabling 'gn check' on webrtc/video. I disabled the check on "video_tests" because it pulls "//webrtc/media/rtc_unittest_main" as a dependency and it defines the _main (that is already defined by "//webrtc/test:test_main"). I will file a bug to solve this in another CL. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2832063003 Cr-Commit-Position: refs/heads/master@{#17859}
2017-04-25 00:35:35 -07:00
"../media:rtc_media_tests_utils",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../media:stream_params",
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort The JsepTransport2 is renamed to JsepTransport. NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
2018-04-13 16:44:34 -07:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:candidate_pair_interface",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../p2p:dtls_transport",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_factory",
"../p2p:dtls_transport_internal",
Create visible fake_ice_transport target Enables downstream projects to use the existing fake ice transport implementation, without taking dependency on gunit Bug: None Change-Id: I78bac9d40aa6e12b55e86f0460bcd98d85c7f214 Reviewed-on: https://webrtc-review.googlesource.com/c/120445 Commit-Queue: Peter Slatala <psla@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26456}
2019-01-29 13:35:05 -08:00
"../p2p:fake_ice_transport",
Factor out the fake port allocator in build. Bug: None Change-Id: I7d757ff33c87ec10c1d1699db36655e67f9e3e73 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133764 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27723}
2019-04-21 22:08:26 -07:00
"../p2p:fake_port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
enabling `gn check` on the whole WebRTC repo BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2918803002 Cr-Commit-Position: refs/heads/master@{#18390}
2017-06-01 13:01:48 -07:00
"../p2p:p2p_test_utils",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
Clean up p2p:rtc_p2p target removing the webrtc need for having sources in it. BUG=webrtc:42226155 Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43540}
2024-12-10 09:26:57 -08:00
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:transport_description",
Add payload type assignment to offer/answer generation. This adds payload types to the codecs at the time when offer is being generated, if they are unassigned at that point. Bug: webrtc:360058654 Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43033}
2024-09-16 20:28:14 +00:00
"../p2p:transport_description_factory",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:gunit_helpers",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:net_helper",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../rtc_base:network_route",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_tests_utils",
Move DTLS related code from p2p/base to p2p/dtls BUG=webrtc:367395350 Change-Id: I3fd1551f974705ce6b10e2c757f4d406a520a2c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370460 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43528}
2024-12-08 21:26:34 -08:00
"../rtc_base:socket",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Replace Invoke in tests with SendTask test helper Bug: webrtc:11318 Change-Id: I14e3fbc694d41c785a61c88d8207005c681576c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271540 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37774}
2022-08-12 15:55:11 +02:00
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base/containers:flat_set",
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback Instead of using PacketTransportInternal::SignalReadPacket. Bug: webrtc:15368 Change-Id: Icdc2d7f85df6db944f0ba0232891e6c5a8986a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340440 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41823}
2024-02-27 09:01:15 +01:00
"../rtc_base/network:received_packet",
Move sigslot to proper third_party directory Extract sigslot into separate target and move it to proper third_party directory. Bug: webrtc:8366 Change-Id: Id2e0712bd020bfad811947803c94553dce06d976 Reviewed-on: https://webrtc-review.googlesource.com/84141 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 15:04:28 +02:00
"../rtc_base/third_party/sigslot",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:metrics",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:explicit_key_value_config",
Move RunLoop test class to its own build target To make it usable in tests without depending on all of CallTest. Bug: None Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36932}
2022-05-19 16:06:23 +02:00
"../test:run_loop",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:scoped_key_value_config",
Delete rtc_base/unittest_main.cc Usage replaced with test/test_main.cc. Bug: webrtc:5996 Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28010}
2019-05-21 13:01:58 +02:00
"../test:test_main",
Introduce RtpTransportInternal and SrtpTransport. SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
2017-07-19 13:38:02 -07:00
"../test:test_support",
Use Abseil container algorithms in pc/ Bug: None Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e Reviewed-on: https://webrtc-review.googlesource.com/c/119862 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26433}
2019-01-28 17:25:26 -08:00
"//third_party/abseil-cpp/absl/algorithm:container",
Migrate pc/ to absl::AnyInvocable based TaskQueueBase interface Bug: webrtc:14245 Change-Id: I9043aa507421a93f0d7ba7406e237f727999b696 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268121 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37478}
2022-07-07 10:08:49 +02:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace std::string::find() == 0 with absl::StartsWith. Bug: None Change-Id: I070c4a5d19455f3a5c5d3ccc05f418545c351987 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172584 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30960}
2020-04-01 12:03:11 +02:00
"//third_party/abseil-cpp/absl/strings",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
]
build: add options to configure libsrtp for boringssl or other libraries Depends on https://webrtc-review.googlesource.com/c/src/+/359928 BUG=webrtc:42234521,webrtc:42224104 Change-Id: I0d6335aa5fb3f090c781bed234ed34d6c98ec299 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359928 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42857}
2024-08-22 11:52:33 -07:00
include_dirs = []
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
if (rtc_build_libsrtp) {
build: add options to configure libsrtp for boringssl or other libraries Depends on https://webrtc-review.googlesource.com/c/src/+/359928 BUG=webrtc:42234521,webrtc:42224104 Change-Id: I0d6335aa5fb3f090c781bed234ed34d6c98ec299 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359928 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42857}
2024-08-22 11:52:33 -07:00
include_dirs += [ "//third_party/libsrtp/srtp" ]
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
deps += [ "//third_party/libsrtp" ]
build: add options to configure libsrtp for boringssl or other libraries Depends on https://webrtc-review.googlesource.com/c/src/+/359928 BUG=webrtc:42234521,webrtc:42224104 Change-Id: I0d6335aa5fb3f090c781bed234ed34d6c98ec299 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359928 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42857}
2024-08-22 11:52:33 -07:00
if (!rtc_build_ssl) {
configs += [ "..:external_ssl_library" ]
}
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
}
if (is_android) {
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
use_default_launcher = false
Create a custom test launcher for android Set use_default_launcher=false in rtc_test on android Bug: webrtc:42223878 Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921 Owners-Override: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43505}
2024-12-04 15:54:49 +01:00
deps += [ "//build/android/gtest_apk:native_test_instrumentation_test_runner_java" ]
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
}
}
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("peerconnection_perf_tests") {
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
testonly = true
Replace RTC_FALLTHROUGH with ABSL_FALLTHROUGH_INTENTED Bug: None Change-Id: I7287403f3fb13b8e30f92ca3cf1882b03bb53a6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166176 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30283}
2020-01-16 14:41:10 +01:00
sources = [ "peer_connection_rampup_tests.cc" ]
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
deps = [
":pc_test_utils",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":peer_connection",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
":peerconnection_wrapper",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:audio_options_api",
Move webrtc::CreatePeerConnectionFactory definition next to decl. This CL moves webrtc::CreatePeerConnectionFactory definitions out of pc:create_pc_factory and merges it with its declaration in the api/ directory. In order to avoid circular dependencies a new build target is created: * api:create_peerconnection_factory Bug: webrtc:9862 Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af Reviewed-on: https://webrtc-review.googlesource.com/c/111186 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25744}
2018-11-22 09:00:13 +01:00
"../api:create_peerconnection_factory",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../api:libjingle_peerconnection_api",
New build target api:media_interface Bug: webrtc:8733 Change-Id: I84bbefb1a5ef8e592db29b79499d60ac80c23464 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153180 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29234}
2019-09-17 14:30:41 +02:00
"../api:media_stream_interface",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api:rtc_error",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../api:rtc_stats_api",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/audio:audio_mixer_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/audio_codecs:audio_codecs_api",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
Migrate PeerConnectionRampUpTest on new perf metrics export API Bug: b/246095034 Change-Id: I3133f6f7517cc303eeec2e860614e91b2ce4402b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276630 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38207}
2022-09-25 01:13:10 +02:00
"../api/test/metrics:global_metrics_logger_and_exporter",
"../api/test/metrics:metric",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/video_codecs:video_codecs_api",
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. Bug: webrtc:13573 Change-Id: Iac9d0eb56a5cae73b75188b9fb257d6815e04bef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299701 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39733}
2023-03-31 15:42:38 +02:00
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../media:rtc_media_tests_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:basic_port_allocator",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../p2p:p2p_test_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:gunit_helpers",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../rtc_base:rtc_base_tests_utils",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:socket_address",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:socket_factory",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Replace Invoke in tests with SendTask test helper Bug: webrtc:11318 Change-Id: I14e3fbc694d41c785a61c88d8207005c681576c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271540 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37774}
2022-08-12 15:55:11 +02:00
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../system_wrappers",
A new PeerConnection level perf test. This test creates a one way audio and video call, allows for bandwidth estimation to ramp up and then runs the call for 10 seconds. The average bandwidth estimate over this time is recorded as a perf metric. This is done at the PeerConnection level with the intention to catch regressions related to ICE configurations. Stats are taken from PeerConnection for BWE, and the network simulation is done with a VirtualSocketServer. Bug: webrtc:7668 Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db Reviewed-on: https://webrtc-review.googlesource.com/78361 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-22 15:40:16 -07:00
"../test:test_support",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("peerconnection_wrapper") {
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
testonly = true
sources = [
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
]
deps = [
":pc_test_utils",
Break out sdp_utils from peerconnection target Bug: webrtc:13634 Change-Id: Ibf337e93ac0581ebaf83b3f5f978704ae73ce9dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35896}
2022-02-03 07:56:22 +00:00
":sdp_utils",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../api:function_view",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
"../api:libjingle_peerconnection_api",
New build target api:media_interface Bug: webrtc:8733 Change-Id: I84bbefb1a5ef8e592db29b79499d60ac80c23464 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153180 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29234}
2019-09-17 14:30:41 +02:00
"../api:media_stream_interface",
Move rtc_error.{h,cc} to its own build target. Bug: webrtc:8733 Change-Id: Idd34d9a88ae62a01b9ea50719872f8188069211e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152320 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29148}
2019-09-10 10:51:23 +02:00
"../api:rtc_error",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
"../api:rtc_stats_api",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Move peerconnectionwrapper.(h|cc) into separate build target Bug: webrtc:10138 Change-Id: I32f8b9721c37075e355b90c3794a4bef6bd46761 Reviewed-on: https://webrtc-review.googlesource.com/c/114548 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26027}
2018-12-17 14:22:06 +01:00
"../test:test_support",
]
}
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
rtc_test("slow_peer_connection_unittests") {
testonly = true
sources = [ "slow_peer_connection_integration_test.cc" ]
deps = [
":integration_test_helpers",
":pc_test_utils",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:dtmf_sender_interface",
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
"../api:libjingle_peerconnection_api",
"../api:scoped_refptr",
"../api/units:time_delta",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
"../p2p:p2p_server_utils",
"../p2p:p2p_test_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
"../test:test_main",
"../test:test_support",
[SourceTracker] Move state to the worker thread, remove mutex. This is in preparation of using the state that SourceTracker manages for more things than only getContributingSources. Audio levels reported via getStats(), aren't consistent with levels reported via getCS. Since more operations will be derived from the ST owned data, moving the management of it away from the audio thread, reduces the potential of contention. Bug: webrtc:14029, webrtc:7517, webrtc:15119 Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39943}
2023-04-25 08:53:42 +02:00
"../test/time_controller:time_controller",
Create a "slow peerconnection unittests" target This CL moves all tests that take more than 5 seconds into the new target. Bug: webrtc:14025 Change-Id: I760d1a270b399b581f41606647740466f6b87e7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36782}
2022-05-05 13:21:19 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
]
}
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
Add generation of CCFB in offer (RFC8888 support) Bug: webrtc:42225697 Change-Id: I288a808c4aae852212b09fdf9551e20fe3066e39 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367205 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43341}
2024-10-31 16:35:03 +00:00
"congestion_control_integrationtest.cc",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"data_channel_integrationtest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"data_channel_unittest.cc",
"dtmf_sender_unittest.cc",
"ice_server_parsing_unittest.cc",
Remove SetLatency/GetLatency from MediaSourceInterface API level Bug: webrtc:10287 Change-Id: I74fad31db98b75791085688438064f9510b0b6fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133165 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27692}
2019-04-18 17:49:49 +02:00
"jitter_buffer_delay_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"jsep_session_description_unittest.cc",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
"legacy_stats_collector_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"local_audio_source_unittest.cc",
"media_stream_unittest.cc",
[Adaptation] Add ability to inject resources on the PeerConnection. This unblocks injecting platform-specific resources, such as power usage signals in Chrome. This CL adds AddAdaptationResource to PeerConnectionInterface and integration tests verifying that if an injected resource is overusing, resolution will soon be reduced. To aid testing, some testing-only classes have been updated. Bug: webrtc:11525 Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31505}
2020-06-11 12:26:53 +02:00
"peer_connection_adaptation_integrationtest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"peer_connection_bundle_unittest.cc",
"peer_connection_crypto_unittest.cc",
"peer_connection_data_channel_unittest.cc",
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-06 15:27:33 +02:00
"peer_connection_encodings_integrationtest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"peer_connection_end_to_end_unittest.cc",
"peer_connection_factory_unittest.cc",
Add tests for advertising dependency descriptor rtp header extension. Bug: webrtc:10342 Change-Id: Ic626fa1c3c8abe13ea2a0dd9b9512043edcef760 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272801 Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38120}
2022-08-25 21:35:30 +09:00
"peer_connection_field_trial_tests.cc",
RtpTransceiverInterface: add header_extensions_to_offer() This change adds exposure of a new transceiver method for getting the total set of supported extensions stored as an attribute, and their direction. If the direction is kStopped, the extension is not signalled in Unified Plan SDP negotiation. Note: SDP negotiation is not modified by this change. Changes: - RtpHeaderExtensionCapability gets a new RtpTransceiverDirection, indicating either kStopped (extension available but not signalled), or other (extension signalled). - RtpTransceiver gets the new method as described above. The default value of the attribute comes from the voice and video engines as before. https://chromestatus.com/feature/5680189201711104. go/rtp-header-extension-ip Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk Bug: chromium:1051821 Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30800}
2020-03-16 13:40:51 +01:00
"peer_connection_header_extension_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"peer_connection_histogram_unittest.cc",
"peer_connection_ice_unittest.cc",
"peer_connection_integrationtest.cc",
"peer_connection_interface_unittest.cc",
"peer_connection_jsep_unittest.cc",
"peer_connection_media_unittest.cc",
"peer_connection_rtp_unittest.cc",
"peer_connection_signaling_unittest.cc",
Reland "Enabling Simulcast use via AddTransceiver." This is a reland of ce470aab518f067a67aa03aaab1fc61a45afa0ec Original change's description: > Enabling Simulcast use via AddTransceiver. > > This change removes the restriction on multiple send encodings when > calling AddTransceiver. If RIDs are not provided in the > simulcast scenario, they are auto-generated by the platform. > > This effectively enables the use of spec-compliant simulcast. > Tests are also added to verify simulcast functionality. > > Bug: webrtc:10075 > Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2 > Reviewed-on: https://webrtc-review.googlesource.com/c/121389 > Reviewed-by: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Amit Hilbuch <amithi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26590} Bug: webrtc:10075 Change-Id: Ib4392e36d5d60b966ecff56d4df69bf60d13a73d Reviewed-on: https://webrtc-review.googlesource.com/c/122962 Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26694}
2019-02-06 17:09:52 -08:00
"peer_connection_simulcast_unittest.cc",
SVC: Check scalability in AddTransceiver and SetParameters ScalabilityMode should be validated against the currently allowed codecs or the currently used codec. Bug: webrtc:11607 Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38433}
2022-10-18 17:05:58 +02:00
"peer_connection_svc_integrationtest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
"proxy_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"rtc_stats_collector_unittest.cc",
"rtc_stats_integrationtest.cc",
"rtc_stats_traversal_unittest.cc",
"rtp_media_utils_unittest.cc",
"rtp_parameters_conversion_unittest.cc",
"rtp_sender_receiver_unittest.cc",
"rtp_transceiver_unittest.cc",
"sctp_utils_unittest.cc",
Return proxied object in OnTransceiver This makes it possible to invoke methods on the transceiver object from any thread. Also makes a few of the mock observer objects thread-safe, to allow testing when the main thread is not the signaling thread. Bug: webrtc:13183 Change-Id: Ic97efef71a21c3075700a028103061032f8d2bcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232120 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35010}
2021-09-16 08:59:11 +00:00
"sdp_offer_answer_unittest.cc",
Rename simulcast SDP serializer which is not a generic SDP serializer but only deals with the simulcast SDP. BUG=None Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40761}
2023-09-14 17:23:59 +02:00
"simulcast_sdp_serializer_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"test/fake_audio_capture_module_unittest.cc",
"test/test_sdp_strings.h",
"track_media_info_map_unittest.cc",
VideoRtpReceiver: factor out VideoRtpTrackSource. This change factors out VideoRtpTrackSource in preparation of building the class out. Bug: chromium:1013590 Change-Id: I015e285b9fcc10b39428dea9f74e0e8648385f62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159925 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29870}
2019-11-22 10:34:37 +01:00
"video_rtp_track_source_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"video_track_unittest.cc",
"webrtc_sdp_unittest.cc",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
]
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
deps = [
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":audio_rtp_receiver",
":audio_track",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Add unit test framework for DataChannelController This is in pursuit of an issue with another CL, but large enough to be worth submitting separately. Bug: webrtc:13931 Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36833}
2022-05-10 08:44:48 +00:00
":data_channel_controller_unittest",
Break up rtc_pc_base: Second round, yellow and orange Bug: webrtc:13805 Change-Id: Ic3560673abc8b27507ebac550576814776688138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256001 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36237}
2022-03-17 13:13:51 +00:00
":dtls_srtp_transport",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":dtls_transport",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
":dtmf_sender",
Migrate PeerConnection tests to EnableMedia api Add test helper to inject fake media engine for those tests. Bug: webrtc:15574 Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 11:32:24 +01:00
":enable_fake_media",
Break out ice_server_parsing from peerconnection target Drive-by: Add empty dummy targets for all the things left in the peerconnection target. They should all move out. Bug: webrtc:13634 Change-Id: I93b193804668decf5feee2a8847403466330e128 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35870}
2022-02-01 13:37:08 +00:00
":ice_server_parsing",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":integration_test_helpers",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":jitter_buffer_delay",
Rename StatsCollector to LegacyStatsCollector. We should have done this a long time ago. Let's do the same for stats_types.h in a separate CL because that file is part of the api/ folder and needs some special care (typedefs and temporarily include helper to avoid breaking downstream projects). Bug: webrtc:14180 Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37426}
2022-07-04 14:36:37 +02:00
":legacy_stats_collector",
Break out local_audio_source from peerconnection target Bug: webrtc:13634 Change-Id: I402e5334869c69b8e7436f3402b04a93f8df7135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250201 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35887}
2022-02-02 11:26:03 +00:00
":local_audio_source",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
":media_protocol_names",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
":media_stream",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":peer_connection",
":peer_connection_factory",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":peer_connection_proxy",
Move proxies into pc/. Bug: webrtc:12787 Change-Id: Ia244d9d22d35436a02cf5a448bd1520cb66ff352 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220321 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34141}
2021-05-26 18:56:30 +02:00
":proxy",
Break out rtc_stats_collector and rtc_stats_traversal Bug: webrtc:13634 Change-Id: I94f8cfe5511ff0798ec4a5ad7df28f3e7b98165d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251241 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35945}
2022-02-08 07:52:10 +00:00
":rtc_stats_collector",
":rtc_stats_traversal",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":rtp_media_utils",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
":rtp_parameters_conversion",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
":rtp_receiver",
":rtp_sender",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":rtp_sender_proxy",
Break out some more pc/ compile targets Bug: webrtc:11967 Change-Id: I98f1fc24174ca3dc82c69ce8092f543ab172a095 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186843 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32356}
2020-10-08 16:35:18 +00:00
":rtp_transceiver",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":rtp_transport_internal",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
":sctp_data_channel",
rtc_pc_base breakup: Brick red targets This is the third wave of dependencies in the rtc_pc_base breakup. Bug: webrtc:13805 Change-Id: I79122f2f0395864d595fdffcc4dafa8ed09c560d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255960 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36230}
2022-03-17 06:26:03 +00:00
":sctp_transport",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":sctp_utils",
Break out sdp_utils from peerconnection target Bug: webrtc:13634 Change-Id: Ibf337e93ac0581ebaf83b3f5f978704ae73ce9dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35896}
2022-02-03 07:56:22 +00:00
":sdp_utils",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
":session_description",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
":simulcast_description",
Rename simulcast SDP serializer which is not a generic SDP serializer but only deals with the simulcast SDP. BUG=None Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40761}
2023-09-14 17:23:59 +02:00
":simulcast_sdp_serializer",
Break out stream_collection.h from peerconnection target Bug: webrtc:13634 Change-Id: I7062f245b3ae5f1d7f38baee0c5b14be3332e658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251061 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35925}
2022-02-07 06:43:48 +00:00
":stream_collection",
Break out track_media_info_map from peerconnection Bug: webrtc:13634 Change-Id: If0a24738fbbff0b3706c3926e7188b8b6a451cdc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35904}
2022-02-03 19:02:38 +00:00
":track_media_info_map",
Break out remaining level-1 targets from rtc_pc_base Bug: webrtc:13805 Change-Id: I39a28489ff121de57a8476da10d297db823db091 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254822 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36182}
2022-03-10 22:15:42 +00:00
":transport_stats",
Move the PeerConnection's usage pattern concept to its own file. This makes it easier to use it from multiple other modules. Bug: webrtc:11995 Change-Id: Id23843ae4600ebe46aed7465e873d107089fd50b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187347 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32361}
2020-10-09 05:43:53 +00:00
":usage_pattern",
Break out more targets from pc/peerconnection Bug: webrtc:11967 Change-Id: Id66556186cb76de231bb26433e5ff64fddb55d79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187486 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32379}
2020-10-12 07:37:31 +00:00
":video_rtp_receiver",
":video_rtp_track_source",
":video_track",
Break out separate compile targets for various classes This reduces the degree of interdependency among modules related to the PeerConnection class, and makes it easier to isolate inappropriate external dependencies. Bug: webrtc:11967 Change-Id: Id9777a2ab690cc349dd5842a3a95e24478144c71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185882 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32235}
2020-09-29 14:21:47 +00:00
":video_track_source",
Break out webrtc_sdp and jsep_session_description These two files turn out to be entangled with each other. Keeping them together for now. This is a simpler approach than the one attempted in https://webrtc-review.googlesource.com/c/src/+/251060 but leaves cleanup of the relationship to a later work item. Bug: webrtc:13634 Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35942}
2022-02-08 05:56:55 +00:00
":webrtc_sdp",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:array_view",
"../api:audio_options_api",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
Move webrtc::CreatePeerConnectionFactory definition next to decl. This CL moves webrtc::CreatePeerConnectionFactory definitions out of pc:create_pc_factory and merges it with its declaration in the api/ directory. In order to avoid circular dependencies a new build target is created: * api:create_peerconnection_factory Bug: webrtc:9862 Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af Reviewed-on: https://webrtc-review.googlesource.com/c/111186 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25744}
2018-11-22 09:00:13 +01:00
"../api:create_peerconnection_factory",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:dtls_transport_interface",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:dtmf_sender_interface",
Introduce new way to enable media in PeerConnectionFactory instead of requiring to pass in call_factory and media_engine webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies Bug: webrtc:15574 Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41049}
2023-10-31 13:51:16 +01:00
"../api:enable_media",
Add EnableMediaWithDefaults to replace SetMediaEngineDefaults Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults Bug: webrtc:15574 Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41059}
2023-11-01 11:08:20 +01:00
"../api:enable_media_with_defaults",
Fixes use after free error when setting a new FrameEncryptor on ChannelSend. This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
2018-10-25 09:52:57 -07:00
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 14:37:36 +01:00
"../api:function_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api:libjingle_logging_api",
Stop using public_deps in api/. Bug: webrtc:8603 Change-Id: I12f91365fa9a3ed12be1a9658999b6f6aad7e5e3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33220 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21584}
2018-01-11 10:13:56 +01:00
"../api:libjingle_peerconnection_api",
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219}
2022-06-14 15:48:26 +02:00
"../api:make_ref_counted",
New build target api:media_interface Bug: webrtc:8733 Change-Id: I84bbefb1a5ef8e592db29b79499d60ac80c23464 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153180 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29234}
2019-09-17 14:30:41 +02:00
"../api:media_stream_interface",
Deprecate AsyncResolver config fields and remove internal usage. Bug: webrtc:12598 Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 11:07:28 +00:00
"../api:mock_async_dns_resolver",
RtpSenderInterface::SetEncoderSelector This cl/ adds a way of setting an EncoderSelector on a specific RtpSenderInterface. This makes it possible to easily use different EncoderSelector on different streams within the same or different PeerConnections. The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor. Iff a EncoderSelector is set on the RtpSender, it will take precedence over the VideoEncoderFactory::GetEncoderSelector. Bug: webrtc:14122 Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:25:46 +02:00
"../api:mock_encoder_selector",
Reland "Move injection of PacketSocketFactory from PC to PCF" This is a reland of commit 905c3a6c73d293882ef11942066ccda52a9e14d1 Change from previous attempt is between ps#1 and ps#2: Use PeerConnectionFactoryInterface::Options to clear the `network_ignore_mask`. Original change's description: > Move injection of PacketSocketFactory from PC to PCF > > Injection via PeerConnectionDependecies was broken, in not accepting > ownership of the injected object. > > Bug: webrtc:7447, webrtc:14204 > Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#37270} Bug: webrtc:7447, webrtc:14204 Change-Id: Ic78ebec2e88a8c44699015c8c7a44e137f44253a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265982 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37290}
2022-06-21 11:37:29 +02:00
"../api:mock_packet_socket_factory",
api/test: Move MockVideoTrack to its own file for sharing Bug: webrtc:9620 Change-Id: Iebe11d3e481dd8046771ded2608a4c57288cd22d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226325 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34494}
2021-07-17 01:32:40 +02:00
"../api:mock_video_track",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api:packet_socket_factory",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api:priority",
Move rtc_error.{h,cc} to its own build target. Bug: webrtc:8733 Change-Id: Idd34d9a88ae62a01b9ea50719872f8188069211e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152320 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29148}
2019-09-10 10:51:23 +02:00
"../api:rtc_error",
Add utility WaitUntil for testing for an eventual condition This replaces the WaitUntilCondition function that was used in the peer_connection_encodings_integrationtest previously. Along with that it adds tests and improved error message printing. As a drive-by, matchers were added for RTCError as these are the return type of this utility function. Bug: webrtc:381524905 Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43494}
2024-12-04 11:36:08 +00:00
"../api:rtc_error_matchers",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api:rtp_transceiver_direction",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/adaptation:resource_adaptation_api",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../api/audio:audio_mixer_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
"../api/audio:audio_processing_statistics",
Introduce api/crypto/BUILD.gn. No-Try: True Bug: webrtc:8733 Change-Id: I8679735be1e5069e371a9f1115a54e897e09964b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152622 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29186}
2019-09-13 14:42:15 +02:00
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Provide Environment to create an audio encoder in tests Bug: webrtc:343086059 Change-Id: I73a48770ae67e529eb5065e957ea6420dea44975 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354881 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42542}
2024-06-18 11:20:40 +02:00
"../api/environment",
Use Environment to keep peer connection factory field trials in ConnectionContext Bug: webrtc:15656 Change-Id: Ice52fcb9ba54a5d0034b59233ceae4f9cefbceae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328860 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41252}
2023-11-27 15:00:21 +01:00
"../api/environment:environment_factory",
In PeerConnection unittests set TaskQueueFactory explicitly instead of using factories for MediaEngine and RtcEventLog that rely on GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: Ie1135f70f4ae4d047c4d6bf2db61489a663385aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141875 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28328}
2019-06-20 10:19:42 +02:00
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api/task_queue",
In PeerConnection unittests set TaskQueueFactory explicitly instead of using factories for MediaEngine and RtcEventLog that rely on GlobalTaskQueueFactory Bug: webrtc:10284 Change-Id: Ie1135f70f4ae4d047c4d6bf2db61489a663385aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141875 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28328}
2019-06-20 10:19:42 +02:00
"../api/task_queue:default_task_queue_factory",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/transport:datagram_transport_interface",
Reland: Wires up WebrtcKeyValueBasedConfig in media engines. This is a reland of https://webrtc-review.googlesource.com/c/src/+/174261 Patchset 1 contains the old cl (plus a merge conflict fix). Later patchets are bufixes: A PeerConnection can be created without a Call instance (in the case of DataChannel only), so we can't always use that to fetch the current trials. Old CL descritpion: This replaces field_trial:: -based functions from system_wrappers. Field trials are still used as fallback, but injectable trials are now possible. // Since re-land is otherwise unchanged, setting previous reviewers as TBR TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org Bug: webrtc:11926 Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32163}
2020-09-22 11:36:35 +02:00
"../api/transport:field_trial_based_config",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/transport:sctp_transport_factory_interface",
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 15:16:49 +02:00
"../api/transport/rtp:rtp_source",
Fix L1Tx target bitrate bug when the standard API is used. There are now multiple ways to configure VP9 L1Tx: - Legacy API: configure legacy SVC and disable encodings, this gets interpreted as disabling spatial layers (non-standard API hack). - Standard API: configure scalability_mode. This can be done either with a single encoding or multiple encodings. As long as only one encoding is active we get a single L1Tx ssrc, same as legacy API. Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks bitrates was only applied in the legacy API code path, not the standard API code path, despite both code paths configuring L1Tx. The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if `number_of_streams == 1`. This is true in legacy code path but not standard code path. The fix is to look at `numberOfSimulcastStreams == 1` instead, which is set to the correct value regardless of code path used. This CL adds comments documenting the difference between `number_of_streams` and `numberOfSimulcastStreams` to reduce the risk of more mistakes like this in the future. Bug: chromium:1455039, b:279161263 Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40287}
2023-06-15 12:49:26 +02:00
"../api/units:data_rate",
Replacing rtc::TimeDelta with webrtc::TimeDelta. This removes the redundant type and replaces all usages. A slight change in behavior is that we no longer get nanosecond resolution. This should not matter since no current code requires nanosecond resolution. Bug: webrtc:9155 Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2 Reviewed-on: https://webrtc-review.googlesource.com/71163 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23174}
2018-05-08 14:52:22 +02:00
"../api/units:time_delta",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/units:timestamp",
Make VideoBitrateAllocatorFactory injectable. This patch makes VideoBitrateAllocatorFactory injectable by adding to PeerConnectionDependencies instead of allowing it to be overridden using MediaEngine (on PeerConnectionFactory). With this patch VideoBitrateAllocatorFactory is owned by the PeerConnection. WANT_LGTM (examples) : sakal@ WANT_LGTM (api/pc) : steveanton@ Bug: webrtc:10547 Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27654}
2019-04-17 07:38:40 +02:00
"../api/video:builtin_video_bitrate_allocator_factory",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/video:encoded_image",
"../api/video:recordable_encoded_frame",
Fix requested_resolution bug where we get stuck with old restrictions. Normally (scaleResolutionDownBy) restrictions are applied at the source which changes the input frame size which triggers reconfiguration with appropriate scaling factors. But when requested_resolution is used, encoder settings are by definition not relative to the input frame size. In order for restrictions to have an effect, they are applied inside ReconfigureEncoder(): you get the minimum between the requested resolution and the restricted resolution. ReconfigureEncoder() happens when you SetParameters(), but the bug here is that we don't do it again once the restrictions are updated. So if restrictions are 540p when you ask for 720p, you get 540p and after restrictions change to unlimited you're still stuck in 540p. The fix is to also trigger ReconfigureEncoder() inside OnVideoSourceRestrictionsUpdated() when the restricted resolution is changing and a requested_resolution is configured. To ensure reconfiguring the encoder "on the fly" like this does not reset initial frame dropping logic, InitialFrameDropper caring about input frame size changing is made conditional on not using requested_resolution. # Slow purple bots failing but they are not affected by this change. NOTRY=True Bug: webrtc:361477261 Change-Id: I1389aa16cf408b0d14e0b5b6f68c2442db955be9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360200 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42882}
2024-08-23 11:16:37 +02:00
"../api/video:resolution",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../api/video:video_bitrate_allocator_factory",
"../api/video:video_codec_constants",
"../api/video:video_frame",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api/video:video_rtp_headers",
Add scalability mode to RTCOutboundRtpStreamStats stats This is in the webrtc-stats spec at https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode. This adds the scalability mode to CodecSpecificInfo which is used to plumb the modes for each simulcast layer. TBR=orphis@webrtc.org Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals. Bug: webrtc:14730 Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-06 10:09:10 +00:00
"../api/video_codecs:scalability_mode",
[Adaptation] Add ability to inject resources on the PeerConnection. This unblocks injecting platform-specific resources, such as power usage signals in Chrome. This CL adds AddAdaptationResource to PeerConnectionInterface and integration tests verifying that if an injected resource is overusing, resolution will soon be reduced. To aid testing, some testing-only classes have been updated. Bug: webrtc:11525 Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31505}
2020-06-11 12:26:53 +02:00
"../call/adaptation:resource_adaptation_test_utilities",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../common_video",
Pass the RtcEventLog instance to ICE via JsepTransportController. This CL fixes a bug that the RtcEventLog owned by PeerConnection was not passed to P2PTransportChannel after JsepTransportController was introduced to deprecate the legacy TransportController. Bug: webrtc:9337 Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5 Reviewed-on: https://webrtc-review.googlesource.com/79964 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23572}
2018-06-11 20:15:46 -07:00
"../logging:fake_rtc_event_log",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:codec",
"../media:media_channel",
"../media:media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:rid_description",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../media:rtc_data_sctp_transport_internal",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../media:rtc_media_config",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:stream_params",
Introduce AudioProcessingFactory interface This interface allows to delegate construction of AudioProcessing to the PeerConnectionFactory where it can provide propagated field trials Bug: webrtc:369904700 Change-Id: Ie05cd771e4a869fa5f43173e127256800ae0727f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365320 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43233}
2024-10-14 11:33:37 +02:00
"../modules/audio_processing:mocks",
Implement RTCRemoteInboundRtpStreamStats for both audio and video. This implements the essentials of RTCRemoteInboundRtpStreamStats. This includes: - ssrc - transportId - codecId - packetsLost - jitter - localId - roundTripTime https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict* The following members are not implemented because they require more work... - From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded, packetsRepaired, burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate. - From RTCRemoteInboundRtpStreamStats: fractionLost. Bug: webrtc:10455, webrtc:10456 Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28073}
2019-05-27 13:40:25 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:basic_port_allocator",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:connection_info",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:dtls_transport_internal",
Factor out the fake port allocator in build. Bug: None Change-Id: I7d757ff33c87ec10c1d1699db36655e67f9e3e73 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133764 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27723}
2019-04-21 22:08:26 -07:00
"../p2p:fake_port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../p2p:p2p_server_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
"../p2p:transport_description",
Penultimate split-up of rtc_p2p build target This takes the rest of the .cc files out of the rtc_p2p build target, leaving only one entangled target to clean up. Bug: webrtc:15796 Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41676}
2024-02-06 15:24:03 +00:00
"../p2p:transport_info",
Move byte_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic7e912cba1218f1eed794cb8c393ac148106b16c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259310 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36634}
2022-04-19 17:35:04 +02:00
"../rtc_base:byte_buffer",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Reland "Split digest methods from ssl target into digest target" This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4 Original change's description: > Split digest methods from ssl target into digest target > > in an attempt to break up the monolithic ssl target. > > BUG=None > > Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@meta.com> > Cr-Commit-Position: refs/heads/main@{#42249} Bug: webrtc:339300437 Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42304}
2024-05-14 08:35:24 -07:00
"../rtc_base:digest",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:gunit_helpers",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:ip_address",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:mdns_responder_interface",
"../rtc_base:net_helper",
"../rtc_base:network",
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064}
2022-02-23 13:44:59 +00:00
"../rtc_base:network_constants",
[DataChannelController] Associate two methods with the network thread Make DataChannelController's AddSctpDataStream and RemoveSctpDataStream be required to be called on the network thread. This moves blocking calls within those methods over to the SctpDataChannel class instead. For production code there's no functional change in this CL. However, this CL: 1) Introduces an actual dedicated network thread to DataChannelController and SctpDataChannel tests. 2) Removes two data_channel_transport() checks inside DCC that were being done on the wrong thread (signaling) and 3) introduces a network calling block to SctpDataChannel, where more network thread related work needs to be done and can be bundled. (to be done in follow-up CLs). Bug: webrtc:11547 Change-Id: I6787ac395e61d4a25ae3a74a123e3357cbb46b54 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298052 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39688}
2023-03-27 12:39:33 +02:00
"../rtc_base:null_socket_server",
Extend testing of prAnswer - Modify munger to take (mutable) std::unique_ptr<SessionDescriptionInterface> rather than cricket::SessionDescription (that latter is embedded in the former) - For all pranswer test cases, do a final SetRemoteDescription(kAnswer) and check that signaling_state == stable Add new test cases: 1) A test case that only applies it as prAnswer on caller (callee is stable) 2) A test case that "scramble" sdb between prAnswer and Anser. Bug: None Change-Id: Ifedd92ade01ae781a2e59d0569133c486c7093fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360781 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42891}
2024-08-29 10:50:50 +02:00
"../rtc_base:random",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../rtc_base:refcount",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:rtc_base_tests_utils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:rtc_certificate_generator",
Add a round-trip test that stats.toJson output is parseable Bug: webrtc:10173 Change-Id: Icf22901824fc85cc390e9450c480d3b7a728dc34 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165385 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30194}
2020-01-09 12:58:23 +01:00
"../rtc_base:rtc_json",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:socket_address",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Replace Invoke in tests with SendTask test helper Bug: webrtc:11318 Change-Id: I14e3fbc694d41c785a61c88d8207005c681576c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271540 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37774}
2022-08-12 15:55:11 +02:00
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Migrate pc/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I1adc22d2998966958750138e66108cf39a8c3d57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178840 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31654}
2020-07-07 19:08:53 +02:00
"../rtc_base/synchronization:mutex",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base/third_party/sigslot",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:metrics",
Add a flag to control legacy vs spec-compliant scalability mode. The goal of the VP9 simulcast project is that when `scalability_mode` is set, multiple encodings are always interpreted as simulcast, even if VP9 or AV1 is used. This CL makes this so, but only if the flag "WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast. When we are ready to ship we will remove the need to use the field trial, but before we ship this we'll want to revisit if SvcRateAllocator can be updated to support simulcast. (Today if we use SvcRateAllocator when VP9 simulcast is used, all encodings except the first one get bitrate=0, causing the test to fail because media is not flowing on all layers.) For now, a TODO is added. Bug: webrtc:14884 Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340 Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39552}
2023-03-14 09:50:19 +01:00
"../test:field_trial",
[DataChannelInterface] Introduce DataChannelInterface::SendAsync() One problem with the existing Send() method is that it has a return value that is problematic for a fully async implementation. A second problem with Send() is that the return value is bool and not RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError. Also, start deprecating `bool Send()` in favor of `void SendAsync()` and adding `network_safety_` flag for posting async operations to the network thread. This flag also takes over from the `connected_to_transport_` which can now be removed. Bug: webrtc:11547, webrtc:13289 Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39817}
2023-04-11 17:32:34 +02:00
"../test:rtc_expect_death",
Move RunLoop test class to its own build target To make it usable in tests without depending on all of CallTest. Bug: None Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36932}
2022-05-19 16:06:23 +02:00
"../test:run_loop",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:scoped_key_value_config",
Add utility WaitUntil for testing for an eventual condition This replaces the WaitUntilCondition function that was used in the peer_connection_encodings_integrationtest previously. Along with that it adds tests and improved error message printing. As a drive-by, matchers were added for RTCError as these are the return type of this utility function. Bug: webrtc:381524905 Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43494}
2024-12-04 11:36:08 +00:00
"../test:wait_until",
Reland "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps" This is to allow testing without using the singleton sctp library. cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure. Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation. patch 1 contain the original cl. patch 2 modifications Bug: none Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840 Reviewed-by: Taylor <deadbeef@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32024}
2020-08-28 09:15:15 +02:00
"../test/pc/sctp:fake_sctp_transport",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"//testing/gtest",
Use Abseil container algorithms in pc/ Bug: None Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e Reviewed-on: https://webrtc-review.googlesource.com/c/119862 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26433}
2019-01-28 17:25:26 -08:00
"//third_party/abseil-cpp/absl/algorithm:container",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Depend directly on destination targets. Makes 'gn check' happy. Followup to https://webrtc-review.googlesource.com/c/src/+/106820 Bug: webrtc:5876, webrtc:9855 Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b Reviewed-on: https://webrtc-review.googlesource.com/c/108760 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-30 22:23:02 +01:00
"//third_party/abseil-cpp/absl/strings",
Stop using public_deps in pc/. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: If18e5a4d212392bbd9b4e1f9c2f00ee79a2ab348 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29864 Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21139}
2017-12-06 11:23:19 +01:00
]
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
if (is_android) {
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
use_default_launcher = false
deps += [
":android_black_magic",
# We need to depend on this one directly, or classloads will fail for
# the voice engine BuildInfo, for instance.
Create a custom test launcher for android Set use_default_launcher=false in rtc_test on android Bug: webrtc:42223878 Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921 Owners-Override: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43505}
2024-12-04 15:54:49 +01:00
"../sdk/android:libjingle_peerconnection_java",
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
]
shard_timeout = 900
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
}
deps += [
":libjingle_peerconnection",
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-25 18:15:09 -08:00
":pc_test_utils",
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
":rtc_pc",
Promote RtcEventLogOutputFile to api/ Preparation for deleting PeerConnectionInterface::StartRtcEventLog method with a PlatformFile argument. Bug: webrtc:6463 Change-Id: Ia9fa1d99a3d87f3bf193e73382690b782ffea65c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135285 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27879}
2019-05-08 13:53:51 +02:00
"../api:rtc_event_log_output_file",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../api:rtc_stats_api",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
PeerConnection end-to-end test with a non-builtin codec BUG=webrtc:8159 Change-Id: I27f3fc8a6204115c480d70238225b0565c41bd81 Reviewed-on: https://webrtc-review.googlesource.com/6283 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20245}
2017-10-10 23:17:17 +02:00
"../api/audio_codecs:audio_codecs_api",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
Add Opus-only audio codec factories Many WebRTC users need only Opus, and no other audio codecs. This makes it convenient for them to do the right thing. To prove that the new factories work, use them in PeerConnectionEndToEndTest. Bug: webrtc:11130 Change-Id: I2c2450ba0fb33ef3b50da8f6cd325cad6b1e59a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160648 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29921}
2019-11-26 14:00:41 +01:00
"../api/audio_codecs:opus_audio_decoder_factory",
"../api/audio_codecs:opus_audio_encoder_factory",
PeerConnection end-to-end test with a non-builtin codec BUG=webrtc:8159 Change-Id: I27f3fc8a6204115c480d70238225b0565c41bd81 Reviewed-on: https://webrtc-review.googlesource.com/6283 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20245}
2017-10-10 23:17:17 +02:00
"../api/audio_codecs/L16:audio_decoder_L16",
"../api/audio_codecs/L16:audio_encoder_L16",
Built in video codec factories. To prepare for making the software codecs optional and injectable, these codec factories provide a way to pass in identical factories as were the default old behaviour. Bug: webrtc:7925 Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921 Reviewed-on: https://webrtc-review.googlesource.com/71162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23096}
2018-05-03 11:28:29 +02:00
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. Bug: webrtc:13573 Change-Id: I69e500f2a18b735396cd00e1ab925243f1a807a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299702 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39744}
2023-03-31 16:49:14 +02:00
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
Reland "Rewrite WebRtcSession media tests as PeerConnection tests" This is a reland of 3df5dcac9b339ba4d3f4969602f094c2c8035b51 Original change's description: > Rewrite WebRtcSession media tests as PeerConnection tests > > Bug: webrtc:8222 > Change-Id: I782a3227e30de70eb8f6c26a48723cb3510a84ad > Reviewed-on: https://webrtc-review.googlesource.com/6640 > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20364} Bug: webrtc:8222 Change-Id: I0a5398170d469eb9223bc781bfb417a85a72a2d2 Reviewed-on: https://webrtc-review.googlesource.com/14380 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20377}
2017-10-20 15:30:51 -07:00
"../call:call_interfaces",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../media:rtc_audio_video",
Enabling 'gn check' on webrtc/video. I disabled the check on "video_tests" because it pulls "//webrtc/media/rtc_unittest_main" as a dependency and it defines the _main (that is already defined by "//webrtc/test:test_main"). I will file a bug to solve this in another CL. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2832063003 Cr-Commit-Position: refs/heads/master@{#17859}
2017-04-25 00:35:35 -07:00
"../media:rtc_media_tests_utils",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/audio_processing",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../p2p:p2p_test_utils",
Removing warning suppression flags from pc/. Bug: webrtc:9251 Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f Reviewed-on: https://webrtc-review.googlesource.com/86820 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23838}
2018-07-03 12:53:23 +02:00
"../rtc_base:safe_conversions",
Fix Chromium style checker warnings for MockAudioDecoder BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2934143003 Cr-Commit-Position: refs/heads/master@{#18587}
2017-06-14 03:31:17 -07:00
"../test:audio_codec_mocks",
Delete rtc_base/unittest_main.cc Usage replaced with test/test_main.cc. Bug: webrtc:5996 Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28010}
2019-05-21 13:01:58 +02:00
"../test:test_main",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../test:test_support",
Create //webrtc/api:libjingle_peerconnection_api + refactorings. Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 04:56:25 -08:00
]
}
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
Add unit test framework for DataChannelController This is in pursuit of an issue with another CL, but large enough to be worth submitting separately. Bug: webrtc:13931 Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36833}
2022-05-10 08:44:48 +00:00
rtc_library("data_channel_controller_unittest") {
testonly = true
sources = [ "data_channel_controller_unittest.cc" ]
deps = [
":data_channel_controller",
":pc_test_utils",
":peer_connection_internal",
Safeguard SctpDataChannel against detached controller Since the lifetime of an SctpDataChannel is not strictly controlled by its controller, the controller might go away before the channel does. This CL guards against this. Bug: webrtc:13931 Change-Id: I07046fe896d1a66bf89287429beb0587382a13a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36852}
2022-05-11 09:35:36 +00:00
":sctp_data_channel",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
[DataChannelController] Associate two methods with the network thread Make DataChannelController's AddSctpDataStream and RemoveSctpDataStream be required to be called on the network thread. This moves blocking calls within those methods over to the SctpDataChannel class instead. For production code there's no functional change in this CL. However, this CL: 1) Introduces an actual dedicated network thread to DataChannelController and SctpDataChannel tests. 2) Removes two data_channel_transport() checks inside DCC that were being done on the wrong thread (signaling) and 3) introduces a network calling block to SctpDataChannel, where more network thread related work needs to be done and can be bundled. (to be done in follow-up CLs). Bug: webrtc:11547 Change-Id: I6787ac395e61d4a25ae3a74a123e3357cbb46b54 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298052 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39688}
2023-03-27 12:39:33 +02:00
"../rtc_base:null_socket_server",
Remove DataChannelController::sctp_data_channels_to_free_ Instead, just use the posted task to release the reference to a pending data channel object. Bug: none Change-Id: I34f0bfd604cab88587a892eaa218856c890fc907 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296767 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39527}
2023-03-09 23:52:43 +01:00
"../test:run_loop",
Add unit test framework for DataChannelController This is in pursuit of an issue with another CL, but large enough to be worth submitting separately. Bug: webrtc:13931 Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36833}
2022-05-10 08:44:48 +00:00
"../test:test_support",
]
}
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
if (is_android) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("android_black_magic") {
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
# The android code uses hacky includes to ssl code. Having this in a
# separate target enables us to keep the peerconnection unit tests clean.
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
testonly = true
sources = [
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"test/android_test_initializer.cc",
"test/android_test_initializer.h",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
]
deps = [
Create a custom test launcher for android Set use_default_launcher=false in rtc_test on android Bug: webrtc:42223878 Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921 Owners-Override: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43505}
2024-12-04 15:54:49 +01:00
"../modules/utility:utility",
"../rtc_base:checks",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Use WebRTC's Java VM initialization in tests. WebRTC should not depend on chromium's //base. Bug: webrtc:13662 Change-Id: Ie660aa0f2477cc747830bba611aa23ed5e732385 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256364 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36581}
2022-04-19 15:32:37 +02:00
"../sdk/android:internal_jni",
Clean up dependencies of peerconnection_unittest. There were a number of unused includes and undeclared dependencies. I removed the includes that were causing problems and added dependencies for the includes that turned out to be needed. Bug: webrtc:7239,webrtc:6828 Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284 Reviewed-on: https://webrtc-review.googlesource.com/1185 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 09:04:28 +02:00
"../sdk/android:libjingle_peerconnection_jni",
]
}
}
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
rtc_library("integration_test_helpers") {
testonly = true
sources = [
"test/integration_test_helpers.cc",
"test/integration_test_helpers.h",
]
deps = [
":audio_rtp_receiver",
":audio_track",
":dtmf_sender",
":jitter_buffer_delay",
Break out local_audio_source from peerconnection target Bug: webrtc:13634 Change-Id: I402e5334869c69b8e7436f3402b04a93f8df7135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250201 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35887}
2022-02-02 11:26:03 +00:00
":local_audio_source",
Finish removal of source files from rtc_pc_base No-try: True Bug: webrtc:13805 Change-Id: Ib7048205fe62379d1a5c01cdbca81ba93b41cf47 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256022 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36355}
2022-03-28 12:06:34 +00:00
":media_session",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":media_stream",
":pc_test_utils",
Break out peer_connection_factory and peer_connection The peerconnection target now has no files, which means that no target that needs .h files depends on it. This is good. Bug: webrtc:13634 Change-Id: I9f194fb224e52a5824eb00f216461c7f928b0308 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251325 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35961}
2022-02-09 05:52:11 +00:00
":peer_connection",
":peer_connection_factory",
Break out -proxy targets from rtc_pc_base Bug: webrtc:13805 Change-Id: Ib9a126c1a12ee8f4a8ba7f888224a862643cabcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254580 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36169}
2022-03-10 10:09:51 +00:00
":peer_connection_proxy",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":remote_audio_source",
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174}
2022-03-10 13:21:28 +00:00
":rtp_media_utils",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_sender",
":rtp_transceiver",
Break out pc/session_description in its own build target (part 1) As a side effect, break out pc/simulcast_description. Step 1: Don't move the {h,cc} files; just declare the targets so that downstream projects can add dependencies on it. Bug: webtc:11967 Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34254}
2021-06-09 09:33:56 +00:00
":session_description",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":usage_pattern",
":video_rtp_receiver",
":video_rtp_track_source",
":video_track",
":video_track_source",
"../api:array_view",
"../api:audio_options_api",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:candidate",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:create_peerconnection_factory",
Migrate PeerConnectionIntegrationWrapper to EnableMedia api Bug: webrtc:15574 Change-Id: I164916b6ba9d29519660b119ed38580c478ea7f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325528 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41064}
2023-11-01 14:43:07 +01:00
"../api:enable_media_with_defaults",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:function_view",
Remove the last build cycle in WebRTC This CL removes the last "nogncheck" comment that was related to a known build cycle. The remaining ones are because of conditional dependencies. Bug: webrtc:8733 Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-20 06:35:28 +00:00
"../api:ice_transport_interface",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"../api:make_ref_counted",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:media_stream_interface",
Deprecate AsyncResolver config fields and remove internal usage. Bug: webrtc:12598 Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40627}
2023-08-25 11:07:28 +00:00
"../api:mock_async_dns_resolver",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:mock_rtp",
"../api:packet_socket_factory",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"../api:sequence_checker",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/audio:audio_mixer_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
Rename AudioProcessingFactory to Builder To stress there is no intention to use each instance more than once. Bug: webrtc:369904700 Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43324}
2024-10-24 16:16:47 +02:00
"../api/audio:builtin_audio_processing_builder",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"../api/metronome",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/transport:field_trial_based_config",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../call/adaptation:resource_adaptation_test_utilities",
"../logging:fake_rtc_event_log",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../media:media_engine",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../media:rtc_media_config",
"../media:rtc_media_tests_utils",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:stream_params",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../modules/audio_processing:audioproc_test_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
Clean up p2p:rtc_p2p target removing the webrtc need for having sources in it. BUG=webrtc:42226155 Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43540}
2024-12-10 09:26:57 -08:00
"../p2p:basic_packet_socket_factory",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:basic_port_allocator",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../p2p:fake_ice_transport",
"../p2p:fake_port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../p2p:p2p_server_utils",
"../p2p:p2p_test_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port",
"../p2p:port_allocator",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:port_interface",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:checks",
Split "helpers" from SSL target to "crypto_random" and rename since it contains helpers mostly related to cryptographically secure random numbers and strings. BUG=webrtc:339300437 Change-Id: I10db939534b25dc792ac1600a4721d1b84521880 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42441}
2024-06-06 07:31:07 -07:00
"../rtc_base:crypto_random",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:gunit_helpers",
"../rtc_base:ip_address",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:mdns_responder_interface",
[SourceTracker] Move state to the worker thread, remove mutex. This is in preparation of using the state that SourceTracker manages for more things than only getContributingSources. Audio levels reported via getStats(), aren't consistent with levels reported via getCS. Since more operations will be derived from the ST owned data, moving the management of it away from the audio thread, reduces the potential of contention. Bug: webrtc:14029, webrtc:7517, webrtc:15119 Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39943}
2023-04-25 08:53:42 +02:00
"../rtc_base:null_socket_server",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:rtc_base_tests_utils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:rtc_certificate_generator",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:rtc_json",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../rtc_base:safe_conversions",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:socket_address",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"../rtc_base:socket_factory",
"../rtc_base:socket_server",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
Split SSL adapters from main ssl build target 1/2 with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
2024-05-17 13:49:05 -07:00
"../rtc_base:ssl_adapter",
Replace Invoke in tests with SendTask test helper Bug: webrtc:11318 Change-Id: I14e3fbc694d41c785a61c88d8207005c681576c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271540 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37774}
2022-08-12 15:55:11 +02:00
"../rtc_base:task_queue_for_test",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
Add tests for FrameBuffer3 and SyncDecoding in PC Integration Tests Change-Id: I04ae6a694599a535347c29706e5ad40cba58155b Bug: webrtc:13672 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251721 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36104}
2022-03-01 10:42:44 +01:00
"../rtc_base/task_utils:repeating_task",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:explicit_key_value_config",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../test:fileutils",
"../test:rtp_test_utils",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:scoped_key_value_config",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../test:test_support",
"../test/pc/sctp:fake_sctp_transport",
[SourceTracker] Move state to the worker thread, remove mutex. This is in preparation of using the state that SourceTracker manages for more things than only getContributingSources. Audio levels reported via getStats(), aren't consistent with levels reported via getCS. Since more operations will be derived from the ST owned data, moving the management of it away from the audio thread, reduces the potential of contention. Bug: webrtc:14029, webrtc:7517, webrtc:15119 Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39943}
2023-04-25 08:53:42 +02:00
"../test/time_controller",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Enable RFC 8888 feedback if negotiated This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated. This should eliminate the need for the "force" flag in the field trial. Bug: webrtc:42225697 Change-Id: Iec7a894c244a417a8499200861550a33f89966a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43398}
2024-11-13 11:54:41 +00:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"//third_party/abseil-cpp/absl/memory",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
]
}
Migrate PeerConnection tests to EnableMedia api Add test helper to inject fake media engine for those tests. Bug: webrtc:15574 Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 11:32:24 +01:00
rtc_library("enable_fake_media") {
testonly = true
visibility = [ ":*" ]
sources = [
"test/enable_fake_media.cc",
"test/enable_fake_media.h",
]
deps = [
":media_factory",
"../api:libjingle_peerconnection_api",
Use Environemnt in MedaFactory::CreateMediaEngine to propagate field trials and task queue factory Bug: webrtc:15656 Change-Id: I2d19e169d2ff1cc871899a0e96b1733333fdc604 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41257}
2023-11-27 17:56:49 +01:00
"../api/environment",
Migrate PeerConnection tests to EnableMedia api Add test helper to inject fake media engine for those tests. Bug: webrtc:15574 Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 11:32:24 +01:00
"../call:call_interfaces",
"../media:rtc_media_tests_utils",
"../rtc_base:checks",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/base:nullability",
Migrate PeerConnection tests to EnableMedia api Add test helper to inject fake media engine for those tests. Bug: webrtc:15574 Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 11:32:24 +01:00
]
}
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
rtc_library("pc_test_utils") {
testonly = true
sources = [
"test/fake_audio_capture_module.cc",
"test/fake_audio_capture_module.h",
Safeguard SctpDataChannel against detached controller Since the lifetime of an SctpDataChannel is not strictly controlled by its controller, the controller might go away before the channel does. This CL guards against this. Bug: webrtc:13931 Change-Id: I07046fe896d1a66bf89287429beb0587382a13a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36852}
2022-05-11 09:35:36 +00:00
"test/fake_data_channel_controller.h",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"test/fake_peer_connection_base.h",
"test/fake_peer_connection_for_stats.h",
"test/fake_periodic_video_source.h",
"test/fake_periodic_video_track_source.h",
"test/fake_rtc_certificate_generator.h",
"test/fake_video_track_renderer.h",
"test/fake_video_track_source.h",
"test/frame_generator_capturer_video_track_source.h",
"test/mock_channel_interface.h",
"test/mock_data_channel.h",
Add unit test framework for DataChannelController This is in pursuit of an issue with another CL, but large enough to be worth submitting separately. Bug: webrtc:13931 Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36833}
2022-05-10 08:44:48 +00:00
"test/mock_peer_connection_internal.h",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"test/mock_peer_connection_observers.h",
"test/mock_rtp_receiver_internal.h",
"test/mock_rtp_sender_internal.h",
Remove all split channels related code Bug: webrtc:13931 Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40376}
2023-06-27 20:07:09 +00:00
"test/mock_voice_media_receive_channel_interface.h",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"test/peer_connection_test_wrapper.cc",
"test/peer_connection_test_wrapper.h",
"test/rtc_stats_obtainer.h",
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-06 15:27:33 +02:00
"test/simulcast_layer_util.cc",
"test/simulcast_layer_util.h",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"test/test_sdp_strings.h",
]
deps = [
Move pc/channel.h to only be used in .cc files This is an implementation API, user classes should in principle only use the channel_interface.h Bug: webrtc:13931 Change-Id: I85c285217858dc087c90a50792e980f731f4439f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260185 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36674}
2022-04-27 12:15:49 +00:00
":channel",
Break out first orange target from rtc_pc_base Bug: webrtc:13805 Change-Id: Ic0618da2cf94ff0c163f2ac72daa2117b0d3b5ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255300 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36187}
2022-03-14 07:16:40 +00:00
":channel_interface",
Migrate PeerConnection tests to EnableMedia api Add test helper to inject fake media engine for those tests. Bug: webrtc:15574 Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41075}
2023-11-03 11:32:24 +01:00
":enable_fake_media",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":jitter_buffer_delay",
":libjingle_peerconnection",
Break out peer_connection_internal from peerconnection target Also make peer_connection_internal not depend on the data_channel_controller (dependency should be in the other direction only). Bug: webrtc:13634 Change-Id: If51358d923348059485928b626a75a1df012ce8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251040 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35926}
2022-02-07 05:07:03 +00:00
":peer_connection_internal",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":rtp_receiver",
":rtp_sender",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
":sctp_data_channel",
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-06 15:27:33 +02:00
":session_description",
":simulcast_description",
Break out stream_collection.h from peerconnection target Bug: webrtc:13634 Change-Id: I7062f245b3ae5f1d7f38baee0c5b14be3332e658 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251061 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35925}
2022-02-07 06:43:48 +00:00
":stream_collection",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
":video_track_source",
"../api:audio_options_api",
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327}
2022-10-07 21:47:49 +00:00
"../api:call_api",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:create_frame_generator",
"../api:create_peerconnection_factory",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
"../api:field_trials_view",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:libjingle_peerconnection_api",
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219}
2022-06-14 15:48:26 +02:00
"../api:make_ref_counted",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:media_stream_interface",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:rtc_error",
"../api:rtc_stats_api",
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616}
2023-04-12 10:45:07 +00:00
"../api:rtp_parameters",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api:scoped_refptr",
"../api:sequence_checker",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/audio:audio_mixer_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/audio_codecs:audio_codecs_api",
Propagate Environment through SimulcastEncoderAdapter when provided Bug: webrtc:15860 Change-Id: Iabd7752ada2f8f774de1e2adc02a4157004bf43c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342720 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41893}
2024-03-12 16:58:18 +01:00
"../api/environment",
Use Environment to keep peer connection factory field trials in ConnectionContext Bug: webrtc:15656 Change-Id: Ice52fcb9ba54a5d0034b59233ceae4f9cefbceae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328860 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41252}
2023-11-27 15:00:21 +01:00
"../api/environment:environment_factory",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
Remove MessageHandler usage in pc test helpers Bug: webrtc:11988 Change-Id: If4175c51b990d1d8ff6eb9a9ba63fa92139b95b6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272404 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37848}
2022-08-19 18:22:15 +02:00
"../api/units:time_delta",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/video:builtin_video_bitrate_allocator_factory",
Add a simulcast test that verifies media is flowing on all layers. Previous tests only asserted that O/A succeeded and that the number of encodings was as expected. This test goes further and also asserts that bytesSent eventually becomes non-zero (after an initial ramp-up time). Let's get testing straight before we add VP9 simulcast support. Bug: webrtc:14885, webrtc:14884 Change-Id: Idccce66698a077264fa0df2c448c8474d2439aea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291960 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39271}
2023-02-08 11:29:20 +01:00
"../api/video:resolution",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
Replace BuiltinVideoDecoderFactory with VideoDecoderFactoryTemplate. Bug: webrtc:13573 Change-Id: Ie933b7902d6b7a548159885178161ed5cb752e22 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299602 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39748}
2023-03-31 13:13:37 +02:00
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. Bug: webrtc:13573 Change-Id: Ie10bdd78dc4c39def83a4dc7806fda3556eb5c2e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299707 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39750}
2023-04-03 15:17:17 +02:00
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../call:call_interfaces",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../media:media_channel",
"../media:media_channel_impl",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../media:rtc_media",
"../media:rtc_media_tests_utils",
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. Bug: webrtc:13573 Change-Id: Ie10bdd78dc4c39def83a4dc7806fda3556eb5c2e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299707 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39750}
2023-04-03 15:17:17 +02:00
"../media:rtc_simulcast_encoder_adapter",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
"../media:video_broadcaster",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../modules/audio_device",
"../modules/audio_processing",
Use parsed packet from RtpTransport::DemuxPacket in engine and call With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused. Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid. Bug: webrtc:7135, webrtc:14795 Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39058}
2023-01-10 14:28:25 +01:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Detangle p2p/connection.cc and port.cc This CL does: - Run IWYU on the relevant elements - Make connection depend on port_interface, not port - Make port_allocator depend only on port - Move some constants from port.h into p2p_constants This allows a dependency graph without ugly groups. Bug: webrtc:15796 Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41618}
2024-01-26 07:11:50 +00:00
"../p2p:connection",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../p2p:fake_port_allocator",
"../p2p:p2p_test_utils",
rtc_p2p: Split turn port and basic port allocator This completes the breakup of the rtc_p2p target. Remaining cleanup is to delete the rtc_p2p target and make clients depend on the base targets. Bug: webrtc:15796 Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41683}
2024-02-06 22:54:25 +00:00
"../p2p:port_allocator",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:rtc_certificate_generator",
"../rtc_base:ssl",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Use WeakPtr in SctpDataChannel. DataChannelController used WeakPtr to clear outstanding references upon destruction - except for the case of SctpDataChannel where we had a pointer+flag for the same purpose. This change updates SctpDataChannel and FakeDataChannelController to use a consistent approach. Bug: none Change-Id: I0248471c241365a2c0de76afbb37302115650194 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295820 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39464}
2023-03-02 15:42:06 +01:00
"../rtc_base:weak_ptr",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:repeating_task",
"../rtc_base/third_party/sigslot",
Move frame_generator_capturer.h|cc to the new target Bug: b/272350185 Change-Id: I3b04e374acb626bec16df22bb63f198b45b790dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300963 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39831}
2023-04-12 15:29:07 +02:00
"../test:frame_generator_capturer",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1 This cl/ 1) move WebRtcKeyValueConfig from api/transport to api/ directory. 2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials). 3) removes usage of webrtc::field_trial:: from the pc/ directory. 4) removes a few unused includes of system_wrappers/field_trial.h. Bug: webrtc:10335 Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36160}
2022-03-09 09:28:10 +01:00
"../test:scoped_key_value_config",
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
"../test:test_support",
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-06 15:27:33 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616}
2023-04-12 10:45:07 +00:00
"//third_party/abseil-cpp/absl/strings",
Rename simulcast flow tests: PeerConnectionEncodingsIntegrationTest. This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
2023-04-06 15:27:33 +02:00
]
Reland "Add a fuzzer test that tries to connect a PeerConnection." This reverts commit ae44fde18854390ca7a51bcab37ef199a1555e38. Reason for revert: Added Chromium compile guards Original change's description: > Revert "Add a fuzzer test that tries to connect a PeerConnection." > > This reverts commit c67b77eee4b08c05638a219723a9141a65015da4. > > Reason for revert: Breaks the libfuzzer chromium bots for WebRTC roll. > > Original change's description: > > Add a fuzzer test that tries to connect a PeerConnection. > > > > Bug: none > > Change-Id: I975c6a4cd5c7dfc4a7689259292ea7d443d270f7 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209182 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33369} > > NOPRESUBMIT=true > > Bug: none > Change-Id: Ib5fa809eb698c64b7c01835e8a311eaf85b19a18 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209640 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33380} Bug: none Change-Id: I07bab58f1216fb91b9b607e7ba978c28838d9411 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210680 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33397}
2021-03-08 07:01:48 +00:00
}
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
svc_tests_resources = [
"../resources/difficult_photo_1850_1110.yuv",
"../resources/photo_1850_1110.yuv",
"../resources/presentation_1850_1110.yuv",
"../resources/web_screenshot_1850_1110.yuv",
]
if (is_ios) {
bundle_data("svc_tests_bundle_data") {
testonly = true
sources = svc_tests_resources
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
}
}
rtc_test("svc_tests") {
sources = [ "test/svc_e2e_tests.cc" ]
data = svc_tests_resources
deps = [
"../api:create_network_emulation_manager",
"../api:create_peer_connection_quality_test_frame_generator",
"../api:create_peerconnection_quality_test_fixture",
"../api:frame_generator_api",
"../api:media_stream_interface",
"../api:network_emulation_manager_api",
"../api:peer_connection_quality_test_fixture_api",
Add scalability mode to RTCOutboundRtpStreamStats stats This is in the webrtc-stats spec at https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode. This adds the scalability mode to CodecSpecificInfo which is used to plumb the modes for each simulcast layer. TBR=orphis@webrtc.org Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals. Bug: webrtc:14730 Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-06 10:09:10 +00:00
"../api:rtc_stats_api",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
"../api:simulated_network_api",
"../api:time_controller",
[PCLF] Fully switch to new metrics export API Bug: b/246095034 Change-Id: I9d588d53320e4eb19cb569db2b97dddc013c22bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276621 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38188}
2022-09-24 16:56:07 +02:00
"../api/test/metrics:global_metrics_logger_and_exporter",
Add a new PeerConnectionE2EQualityTestFixture::AddPeer method. Change-Id: Ic5879613db51a00e3e958931f5eda19fda1ae94a Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282640 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38608}
2022-11-10 15:42:53 +01:00
"../api/test/pclf:media_configuration",
"../api/test/pclf:media_quality_test_params",
"../api/test/pclf:peer_configurer",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
"../api/video_codecs:video_codecs_api",
Apply include-cleaner to api/test/[^/]* e.g all files in the api/test folder not including subdirectories Bug: webrtc:42226242 Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Cr-Commit-Position: refs/heads/main@{#42773}
2024-08-13 05:43:47 -07:00
"../media:media_constants",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
"../modules/video_coding:webrtc_vp9",
Simplify creation of SvcTestParameters in pc/test/svc_e2e_tests.cc No functional changes are intended. Bug: None Change-Id: I361b04da5ed22e12951d8bcc1d16e4e4d00985d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275901 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#38139}
2022-09-21 08:56:40 +09:00
"../modules/video_coding/svc:scalability_mode_util",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
"../rtc_base/containers:flat_map",
"../system_wrappers:field_trial",
"../test:field_trial",
"../test:fileutils",
"../test:test_main",
"../test:test_support",
Create a custom test launcher for android Set use_default_launcher=false in rtc_test on android Bug: webrtc:42223878 Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921 Owners-Override: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43505}
2024-12-04 15:54:49 +01:00
"../test/network:simulated_network",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
"../test/pc/e2e:network_quality_metrics_reporter",
Reland "[DVQA] Create separate BUILD.gn file for video analyzer" This reverts commit 76793c300fdd87fa8fd8be3dd2e5faf8c1916e96. Reason for revert: Can't cleanly revert the old one. A forward fix will be provided. Original change's description: > Revert "[DVQA] Create separate BUILD.gn file for video analyzer" > > This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c. > > Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview > > > Original change's description: > > [DVQA] Create separate BUILD.gn file for video analyzer > > > > Bug: None > > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17 > > No-try: True > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141 > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38662} > > Bug: None > Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000 > Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38672} Bug: None Change-Id: I74506eaa6a1060bf87e651881c86b4f576f447ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38676}
2022-11-18 09:47:40 +00:00
"../test/pc/e2e/analyzer/video:default_video_quality_analyzer",
SVC: Add end to end tests for VP8 and VP9 The tests check that the various scalability mode are supported and the frames are marked properly by the encoder with their spatial and temporal index. The same information is then checked on the receiving side. A new member is added on EncodedImage to store the temporal index, and is filled by the encoders and retreived by the ref finder objects on the decoding side. Bug: webrtc:11607 Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37303}
2022-04-25 17:28:00 +02:00
]
if (is_ios) {
deps += [ ":svc_tests_bundle_data" ]
}
}
GN: Add rtc_pc_unittests BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True TBR=perkj@webrtc.org Review-Url: https://codereview.webrtc.org/2051413003 Cr-Commit-Position: refs/heads/master@{#13107}
2016-06-13 00:08:24 -07:00
}
Reference in New Issue Copy Permalink