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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.
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
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
}
config("rtc_pc_config") {
Remove HAVE_SRTP define and unmaintained code. It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained and failed to compile. BUG=webrtc:7294 Review-Url: https://codereview.webrtc.org/2729373002 Cr-Commit-Position: refs/heads/master@{#17074}
2017-03-06 11:32:22 -08:00
defines = []
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. This allows building without SCTP support (and even building/running tests). The "HAVE_SCTP" define has been functional for a while, but there wasn't any easy way to turn it on/off. NOTRY=True BUG=webrtc:6933 Review-Url: https://codereview.webrtc.org/2593313002 Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 10:53:38 -08:00
if (rtc_enable_sctp) {
Prefix HAVE_SCTP macro with WEBRTC_. Generated automatically with: git grep -l "\bHAVE_SCTP\b" | xargs \ sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g' Bug: webrtc:11142 Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33042}
2021-01-18 13:24:40 +01:00
defines += [ "WEBRTC_HAVE_SCTP" ]
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. This allows building without SCTP support (and even building/running tests). The "HAVE_SCTP" define has been functional for a while, but there wasn't any easy way to turn it on/off. NOTRY=True BUG=webrtc:6933 Review-Url: https://codereview.webrtc.org/2593313002 Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 10:53:38 -08:00
}
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
}
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("rtc_pc_base") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
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
defines = []
sources = [
"channel.cc",
"channel.h",
(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
"channel_interface.h",
"channel_manager.cc",
"channel_manager.h",
"dtls_srtp_transport.cc",
"dtls_srtp_transport.h",
"dtls_transport.cc",
"dtls_transport.h",
"external_hmac.cc",
"external_hmac.h",
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.cc",
"ice_transport.h",
(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.cc",
"jsep_transport.h",
"jsep_transport_controller.cc",
"jsep_transport_controller.h",
"media_session.cc",
"media_session.h",
"rtcp_mux_filter.cc",
"rtcp_mux_filter.h",
"rtp_media_utils.cc",
"rtp_media_utils.h",
"rtp_transport.cc",
"rtp_transport.h",
"rtp_transport_internal.h",
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
"sctp_data_channel_transport.cc",
"sctp_data_channel_transport.h",
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.cc",
"sctp_transport.h",
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
"sctp_utils.cc",
"sctp_utils.h",
(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.cc",
"session_description.h",
"simulcast_description.cc",
"simulcast_description.h",
"srtp_filter.cc",
"srtp_filter.h",
"srtp_session.cc",
"srtp_session.h",
"srtp_transport.cc",
"srtp_transport.h",
"transport_stats.cc",
"transport_stats.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.h",
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
]
deps = [
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
":media_protocol_names",
Securely clear memory containing key information / passwords before freeing. The previously used "memset(ptr, 0, size)" can get optimized away by compilers if "ptr" is not used afterwards. A new class "ZeroOnFreeBuffer" is introduced that can hold sensitive data and that automatically clears underlying memory when it's no longer used. Bug: webrtc:8806, webrtc:8897, webrtc:8905 Change-Id: Iedddddf80790f9af0addaab3346ec5bff102917d Reviewed-on: https://webrtc-review.googlesource.com/41941 Commit-Queue: Joachim Bauch <jbauch@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22334}
2018-03-07 20:02:26 +01:00
"../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",
Move webrtc/audio_*.h to webrtc/api/call BUG=webrtc:5878 NOTRY=True Review-Url: https://codereview.webrtc.org/2059703002 Cr-Commit-Position: refs/heads/master@{#13996}
2016-08-31 07:33:05 -07:00
"../api:call_api",
Take FunctionView rather than any functor reference in the rtc::Thread::Invoke to generate less versions of the function template and FunctorMessageHandler helper thus producing less binary size Bug: None Change-Id: Idbd6fb1e1f23b9b2dc4e4306a74ef11e74ba94cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161044 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29962}
2019-11-29 12:56:43 +01:00
"../api:function_view",
Make ICE transports injectable. Bug: chromium:1024965 Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Honghai Zhang <honghaiz@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29807}
2019-11-15 12:33:05 -08:00
"../api:ice_transport_factory",
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
"../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:packet_socket_factory",
Pass datachannel priority in DC open messages This adds priority to the API configuration of datachannels, and passes the value in the OPEN message. It does not yet influence SCTP prioritization of messages. Bug: chromium:1083227 Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31287}
2020-05-16 08:37:49 +02:00
"../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",
"../api:rtp_parameters",
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",
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",
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",
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:options",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
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 MediaTransportInterface out of the libjingle_peerconnection_api target And move related files into api/transport/ and api/transport/media/. The moved files are unchanged, except that congestion_control_interface.h and datagram_transport_interface.h no longer include media_transport_interface.h, instead, they forward declare the few MediaTransport* types they reference. Bug: webrtc:8733 Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-12 13:59:36 +02:00
"../api/transport:datagram_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/transport:enums",
"../api/transport:sctp_transport_factory_interface",
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",
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:video_bitrate_allocator_factory",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 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@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
[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:call_interfaces",
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",
"../call:rtp_receiver",
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
"../common_video",
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
"../common_video:common_video",
Move datagram_dtls_adaptor from p2p/base/ to pc/ Datagram_dtls_adaptor needs access to rtp_rtcp modules and this moves helps to keep p2p/base/ without dependency on rtp_rtcp. Bug: webrtc:9719 Change-Id: Ic337be3fb9f68106187a84efa815eefbe5b0fcd7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145267 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28533}
2019-07-10 11:07:40 -07:00
"../logging:ice_log",
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
"../media:rtc_data",
Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) Reason for revert: Relanding the orginal CL. The breakage would be a flakey build. Original issue's description: > Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) > > Reason for revert: > The Android 32 (more config) bot is broken. > > Original issue's description: > > Try to fix the binary size increase issue on Chromium. > > > > The target common_video used to depend on rtc_media_base which introduces > > the dependency on p2p. This probably causes the binary size increase on Win > > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly. > > > > BUG=chromium:734631 > > > > Review-Url: https://codereview.webrtc.org/2945233002 > > Cr-Commit-Position: refs/heads/master@{#18693} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9ed16097375fb8d9b45623c58d9086d33e503760 > > TBR=kjellander@webrtc.org,deadbeef@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=chromium:734631 > > Review-Url: https://codereview.webrtc.org/2949953003 > Cr-Commit-Position: refs/heads/master@{#18694} > Committed: https://chromium.googlesource.com/external/webrtc/+/c2e208a9249452590fa282ef5aba43e480bc5794 TBR=kjellander@webrtc.org,deadbeef@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:734631 Review-Url: https://codereview.webrtc.org/2949883003 Cr-Commit-Position: refs/heads/master@{#18695}
2017-06-21 01:02:59 -07:00
"../media:rtc_h264_profile_id",
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
"../media:rtc_media_base",
[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",
Integrate datagram feedback loop This change removes RTCP Feedback loop if we are using datagram transport by removing transport sequence numbers from RTP packets and recreating RTCP Feedback from Datagram ACKs and Timestamps. - For outgoing RTP packets, remove transport sequence number and store it with datagram_id. Note that removing transport sequence numbers does not only save 4-8 bytes per packet, but also prevents generation of feedback packets on the receiver side. - When datagram ACKs, we re-created RTCP feedback with timestamp. - Replacing previous assumption that datagram_id was the same as packet_id by storing incremental counter of datagram ids (I noticed some packets come without packet_id, which is a bit strange, but easy to support and it's also good not to rely on packet_ids being unique across multiple ssrcs). Bug: webrtc:9719 Change-Id: Iecfe938ecea1a74e7c9e1484f0e985d72643d4a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145269 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28542}
2019-07-10 15:44:56 -07:00
"../modules/rtp_rtcp:rtp_rtcp",
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",
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:rtc_p2p",
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",
Rename RoboCaller to CallbackList. As discussed on a design review, the name RoboCaller is not clear enough and switching to CallbackList will provide readability benefits. Bug: webrtc:11943 Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32478}
2020-10-23 12:04:40 +02:00
"../rtc_base:callback_list",
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",
Deprecate owned naked pointers to cricket::SessionDescription Recommended usage is to create copies with Clone() and to call JsepSessionDescription::Initialize using std::move. Bug: webrtc:10612 Change-Id: I626a08a35ba8e112471cec0374c944e96f8effbc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135941 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27911}
2019-05-10 09:31:04 +02:00
"../rtc_base:deprecation",
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_task_queue",
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:socket",
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",
Negotiating Simulcast in the initial offer/answer - Part1. This change adds Simulcast negotiation to media session offers/answers. Next step is to add negotiation logic to PeerConnection. Bug: webrtc:10075 Change-Id: Iea3a1084c16058f0efbc974cf623ec05c3c7a74f Reviewed-on: https://webrtc-review.googlesource.com/c/115790 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26115}
2019-01-02 10:13:58 -08:00
"../rtc_base:stringutils",
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/network:sent_packet",
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",
Change StartAecDump methods to work with FILE* and FileWrapper Bug: webrtc:6463 Change-Id: Id275975decb9b2876021ced19ee9f279b07bea53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140283 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28233}
2019-06-11 14:04:16 +02:00
"../rtc_base/system:file_wrapper",
Add missing RTC_EXPORT for the component build. Bug: webrtc:9419 Change-Id: I3225259fb4cc55e9820f590928795f4587f1e3cb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153884 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29479}
2019-09-23 14:54:28 +02:00
"../rtc_base/system:rtc_export",
In criket::BaseChannel replace AsyncInvoker with task queue functions all invokes, as well as BaseChannel constructor and destructor should run on the same task queue which allow to use simpler cancellation of pending task on BaseChannel destruction Bug: webrtc:12339 Change-Id: I311b6de940cc24cf6bb5b49e1bbd132fea2439e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202032 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33009}
2021-01-15 17:29:53 +01:00
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
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",
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",
Propagate datagram SentNotification for RTCP packets 1. Propagate sent notification for all packets, including RTCP 2. Add Field Trial to disable datagram => RTCP feedback loop translation (for tests and experiments only, because having two feedback loops add extra overhead). Bug: webrtc:9719 Change-Id: Ia3143cc79d127ae331210c86d6675d6e778e962b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145460 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28568}
2019-07-12 16:32:30 -07:00
"../system_wrappers:field_trial",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
Clarify expectation on GlobalLock Merge GlobalLock and GlobalLockPod, make member private. annotate creation of all GlobalLocks with ABSL_CONST_INIT Bug: None Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29447}
2019-10-10 11:12:15 +02:00
"//third_party/abseil-cpp/absl/base:core_headers",
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 _stricmp with absl::EqualsIgnoreCase All uses check only for equality. Bug: webrtc:6424 Change-Id: I8755dde02370c89dbc2226bb703664c9e4f88bdb Reviewed-on: https://webrtc-review.googlesource.com/c/106383 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25274}
2018-10-19 15:29:54 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in pc This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'pc' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12 Reviewed-on: https://webrtc-review.googlesource.com/84181 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 16:47:43 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
GN Templates: Move common_inherited_config to the template. Remove common_inherited_config from the targets and add it to the template instead. BUG=webrtc:6187 NOTRY=True Review-Url: https://codereview.webrtc.org/2311843002 Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 06:10:18 -07:00
public_configs = [ ":rtc_pc_config" ]
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
}
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
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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
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.
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
deps = [
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
":rtc_pc_base",
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
"../media:rtc_audio_video",
]
}
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") {
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",
]
}
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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
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
cflags = []
sources = [
Refactor DataChannel control out of PeerConnection This is step 1-3 of the refactoring process outlined in comment #1 of bugs.webrtc.org/11146 Bug: webrtc:11146 Change-Id: Iccad009bc0585f99d207a6ddb42fd8e71312fc0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161003 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29970}
2019-12-02 09:56:02 +01:00
"data_channel_controller.cc",
Refactoring DataChannelController from PeerConnection part 4 This CL: - Moved HasDataChannel and data_channel_type_ - Moved rtp_data_channels_ - Moved sctp_data_channels_ - Moved data_channel_controller to its own .h file - Various changes to reduce the coupling between the classes - Removed friendship between DataChannelController and PeerConnection Bug: webrtc:11146 Change-Id: Ib8c395e4c90ce34baf40812d1dade0ffa79f2438 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161094 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29987}
2019-12-03 14:04:21 +01:00
"data_channel_controller.h",
Split DataChannel into two separate classes for RTP and SCTP. Done in preparation for some threading changes that would be quite messy if implemented with the class as-is. This results in some code duplication, but is preferable to one class having two completely different modes of operation. RTP data channels are in the process of being removed anyway, so the duplicated code won't last forever. Bug: webrtc:9883 Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-09 15:32:34 -07:00
"data_channel_utils.cc",
"data_channel_utils.h",
(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
"ice_server_parsing.cc",
"ice_server_parsing.h",
"jsep_ice_candidate.cc",
"jsep_session_description.cc",
"local_audio_source.cc",
"local_audio_source.h",
"media_stream_observer.cc",
"media_stream_observer.h",
"peer_connection.cc",
"peer_connection.h",
"peer_connection_factory.cc",
"peer_connection_factory.h",
"peer_connection_internal.h",
"rtc_stats_collector.cc",
"rtc_stats_collector.h",
"rtc_stats_traversal.cc",
"rtc_stats_traversal.h",
Split DataChannel into two separate classes for RTP and SCTP. Done in preparation for some threading changes that would be quite messy if implemented with the class as-is. This results in some code duplication, but is preferable to one class having two completely different modes of operation. RTP data channels are in the process of being removed anyway, so the duplicated code won't last forever. Bug: webrtc:9883 Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-09 15:32:34 -07:00
"rtp_data_channel.cc",
"rtp_data_channel.h",
"sctp_data_channel.cc",
"sctp_data_channel.h",
Refactor webrtc::PeerConnection to split out offer/answer This reduces the size of peer_connection.cc by more than 2000 lines. Design doc for refatoring (available on request): https://docs.google.com/document/d/1ETeUhon9sJihEUpA9ZZHpOGhzDqlZGLQOk3cD_CjKDM/edit Bug: webrtc:11995 Change-Id: I9ed8603807b45bb192a01df026755cb6b5365291 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185801 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32212}
2020-09-28 13:02:07 +00:00
"sdp_offer_answer.cc", # TODO: Make separate target when not circular
"sdp_offer_answer.h", # dependent on peerconnection.h
(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
"sdp_serializer.cc",
"sdp_serializer.h",
"sdp_utils.cc",
"sdp_utils.h",
"stats_collector.cc",
"stats_collector.h",
"stream_collection.h",
"track_media_info_map.cc",
"track_media_info_map.h",
"webrtc_sdp.cc",
"webrtc_sdp.h",
"webrtc_session_description_factory.cc",
"webrtc_session_description_factory.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
]
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",
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
":connection_context",
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",
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",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
":media_protocol_names",
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",
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
":peer_connection_message_handler",
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",
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_pc_base",
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 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",
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_transmission_manager",
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
":sdp_state_provider",
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
":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
":transceiver_list",
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_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",
[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",
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
"../api:call_api",
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:callfactory_api",
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
"../api:fec_controller_api",
Insert frame transformer between Encoded and Packetizer. Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27 16:16:55 +01:00
"../api:frame_transformer_interface",
Make ICE transports injectable. Bug: chromium:1024965 Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Honghai Zhang <honghaiz@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29807}
2019-11-15 12:33:05 -08:00
"../api:ice_transport_factory",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api:libjingle_logging_api",
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",
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",
Add base class NetworkPredictor and NetworkPredictorFactory and wire up. Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe. Bug: webrtc:10492 Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 13:48:24 +02:00
"../api:network_state_predictor_api",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api:packet_socket_factory",
Pass datachannel priority in DC open messages This adds priority to the API configuration of datachannels, and passes the value in the OPEN message. It does not yet influence SCTP prioritization of messages. Bug: chromium:1083227 Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31287}
2020-05-16 08:37:49 +02:00
"../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",
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",
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
"../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",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +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",
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 superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/adaptation:resource_adaptation_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/audio_codecs:audio_codecs_api",
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",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/crypto:options",
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",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
Expose TaskQueueFactory for webrtc::Call in peer connection api making a step for GlobalTaskQueueFactory to be optional way to provide TaskQueueFactory Bug: webrtc:10284 Change-Id: Ife838b3691c256820973118bc5b3cb372dea09cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130488 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27423}
2019-04-01 10:33:16 +02:00
"../api/task_queue",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/transport:bitrate_settings",
Move MediaTransportInterface out of the libjingle_peerconnection_api target And move related files into api/transport/ and api/transport/media/. The moved files are unchanged, except that congestion_control_interface.h and datagram_transport_interface.h no longer include media_transport_interface.h, instead, they forward declare the few MediaTransport* types they reference. Bug: webrtc:8733 Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-12 13:59:36 +02:00
"../api/transport:datagram_transport_interface",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/transport:enums",
Adds injectable trials from peerconnection down to transport controller. This will be immediately useful to guarantee consistent state across components referencing the pacer, but will be a net benefit overall imo. Bug: webrtc:10809 Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29859}
2019-11-15 17:18:52 +01:00
"../api/transport:field_trial_based_config",
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/transport:network_control",
"../api/transport:sctp_transport_factory_interface",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/transport:webrtc_key_value_config",
Control PeerConnectionFactory's default min/starting/max bitrates from experiment Bug: webrtc:10865 Change-Id: Ida88d34d9ee4f390af44d157eef55288fde3773e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148840 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28842}
2019-08-13 10:48:48 +02:00
"../api/units:data_rate",
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",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../api/video:video_bitrate_allocator_factory",
"../api/video:video_codec_constants",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 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@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../api/video_codecs:video_codecs_api",
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
"../call:call_interfaces",
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
"../common_video",
Bind the structured ICE logging with P2PTransportChannel. This change list passes the instance of RtcEventLog from Peerconnection down to P2PTransportChannel, and binds the structured ICE logging with ICE layer objects. Logs of ICE connectivity checks are injected for candidate pairs. TBR=terelius@webrtc.org Bug: None Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821 Reviewed-on: https://webrtc-review.googlesource.com/34660 Commit-Queue: Qingsi Wang <qingsi@google.com> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21884}
2018-01-30 17:13:09 -08:00
"../logging:ice_log",
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
"../media:rtc_data",
"../media:rtc_media_base",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../media:rtc_media_config",
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
"../modules/audio_processing:audio_processing_statistics",
Reland "Adding a restriction for legal RID values." This is a reland of 07f3279a730980583403b78c3762c5d246d1d9be Original change's description: > Adding a restriction for legal RID values. > > According to the spec, RID values should be constrained to only > alpha-numeric values. This was not enforced in our implementation to > allow for more flexibility. > It has been brought to our attention that some values that we currently > consider legal (such as the '~', '=' ';' characters) might cause confusion > with the simulcast syntax that uses these characters to indicate other > meanings. > What's worse, is that some characters, when used in RIDs (such as > \u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some > users which might also be a health hazard. > This change resolves these issues by restricting RIDs to alpha-numeric. > > Bug: webrtc:10491 > Change-Id: I16e262c87525d0289764beacd098e1525a355463 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Amit Hilbuch <amithi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27499} TBR=steveanton@webrtc.org Bug: webrtc:10491 Change-Id: I856581306a9258480ee9184f12b55c2a23dd8636 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131983 Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27530}
2019-04-08 14:11:57 -07:00
"../modules/rtp_rtcp:rtp_rtcp_format",
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:rtc_p2p",
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",
Rename RoboCaller to CallbackList. As discussed on a design review, the name RoboCaller is not clear enough and switching to CallbackList will provide readability benefits. Bug: webrtc:11943 Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32478}
2020-10-23 12:04:40 +02:00
"../rtc_base:callback_list",
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",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../rtc_base:deprecation",
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",
"../rtc_base:network_constants",
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_approved",
Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." This is a reland of 1dddaa1a84330091ca083c950ef2e24a85a48fc8 The regression that caused the original CL to be reverted was the fact that invoking SetLocalDescription() inside of the CreateOffer() callback was no longer executing synchronously and immediately. In this CL, the original CL is patched so that the CreateOffer() operation is marked as completed just before invoking the CreateOffer() callback (versus doing it just afterwards). This ensures that the OperationsChain is popped before the callback runs. The same applies for CreateAnswer(). See diff between Patch Set 1 (Original CL) and the latest Patch Set. Original change's description: > [PeerConnection] Use an OperationsChain in PeerConnection for async ops. > > For background, motivation, requirements and implementation notes, see > https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing > > Using the OperationsChain will unblock future CLs from chaining multiple > operations together such as implementing parameterless > setLocalDescription(). > > In this CL, the OperationsChain is used in existing signaling operations > with little intended side-effects. An operation that is chained onto an > empty OperationsChain will for instance execute immediately, and > SetLocalDescription() and SetRemoteDescription() are implemented as > "synchronous operations". > > The lifetime of the PeerConnection is not indended to change as a result > of this CL: All chained operations use a WeakPtr to the PC to ensure > use-after-free does not happen. > > There is one notable change though: CreateOffer() and CreateAnswer() will > asynchronously delay other signaling methods from executing until they > have completed. > > Drive-by fix: This CL also ensures that early failing > CreateOffer/CreateAnswer operation's observers are invoked if the > PeerConnection is destroyed while a PostCreateSessionDescriptionFailure > is pending. > > Bug: webrtc:11019 > Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29605} TBR=steveanton@webrtc.org Bug: webrtc:11019 Change-Id: I57b4496e63378c91c24679ee496e21f5cb6a0e59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158524 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29630}
2019-10-28 12:09:49 +01:00
"../rtc_base:rtc_operations_chain",
Change PlayoutLatency setLatency zero-threshold value. This is needed to match behaviour described in this spec: https://github.com/henbos/webrtc-timing/pull/2 Bug: webrtc:10287 Change-Id: Idce9af2ec63705dfbfb500b7dbbf755ed3eab571 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131336 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ruslan Burakov <kuddai@google.com> Cr-Commit-Position: refs/heads/master@{#27480}
2019-04-08 11:38:46 +02:00
"../rtc_base:safe_minmax",
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",
"../rtc_base:threading",
Reland "[PeerConnection] Use an OperationsChain in PeerConnection for async ops." This is a reland of 1dddaa1a84330091ca083c950ef2e24a85a48fc8 The regression that caused the original CL to be reverted was the fact that invoking SetLocalDescription() inside of the CreateOffer() callback was no longer executing synchronously and immediately. In this CL, the original CL is patched so that the CreateOffer() operation is marked as completed just before invoking the CreateOffer() callback (versus doing it just afterwards). This ensures that the OperationsChain is popped before the callback runs. The same applies for CreateAnswer(). See diff between Patch Set 1 (Original CL) and the latest Patch Set. Original change's description: > [PeerConnection] Use an OperationsChain in PeerConnection for async ops. > > For background, motivation, requirements and implementation notes, see > https://docs.google.com/document/d/1XLwNN2kUIGGTwz9LQ0NwJNkcybi9oKnynUEZB1jGA14/edit?usp=sharing > > Using the OperationsChain will unblock future CLs from chaining multiple > operations together such as implementing parameterless > setLocalDescription(). > > In this CL, the OperationsChain is used in existing signaling operations > with little intended side-effects. An operation that is chained onto an > empty OperationsChain will for instance execute immediately, and > SetLocalDescription() and SetRemoteDescription() are implemented as > "synchronous operations". > > The lifetime of the PeerConnection is not indended to change as a result > of this CL: All chained operations use a WeakPtr to the PC to ensure > use-after-free does not happen. > > There is one notable change though: CreateOffer() and CreateAnswer() will > asynchronously delay other signaling methods from executing until they > have completed. > > Drive-by fix: This CL also ensures that early failing > CreateOffer/CreateAnswer operation's observers are invoked if the > PeerConnection is destroyed while a PostCreateSessionDescriptionFailure > is pending. > > Bug: webrtc:11019 > Change-Id: I521333e41d20d9bbfb1e721609f2c9db2a5f93a9 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157305 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29605} TBR=steveanton@webrtc.org Bug: webrtc:11019 Change-Id: I57b4496e63378c91c24679ee496e21f5cb6a0e59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158524 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29630}
2019-10-28 12:09:49 +01:00
"../rtc_base:weak_ptr",
Reland "Remove the injectable bitrate allocation strategy API." This is a reland of 80cb3f6db622442b6360e67851e8903aa0d06d03 Original change's description: > Remove the injectable bitrate allocation strategy API. > > This removes PeerConnectionInterface::SetBitrateAllocationStrategy() > plus a ton of now-dead code. > > Bug: webrtc:10556 > Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28523} TBR=kwiberg@webrtc.org Bug: webrtc:10556 Change-Id: Ic17a7a7cc447292306876ee9582ad62fd2499765 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145900 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28585}
2019-07-09 12:31:20 +02:00
"../rtc_base/experiments:field_trial_parser",
Remove superfluous #includes from peer_connection.cc, and add IWYU Also adds a script that runs iwyu to the tools_webrtc directory. Bug: webrtc:11995 Change-Id: I2185a9957e3578c2ec6d0d306061a48fcfe840d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188800 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32431}
2020-10-18 16:51:47 +00:00
"../rtc_base/network:sent_packet",
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",
Change StartAecDump methods to work with FILE* and FileWrapper Bug: webrtc:6463 Change-Id: Id275975decb9b2876021ced19ee9f279b07bea53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140283 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28233}
2019-06-11 14:04:16 +02:00
"../rtc_base/system:file_wrapper",
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",
Export symbols needed by the Chromium component build (part 2). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). Bug: webrtc:9419 Change-Id: I6f27003001548ea9d54412fdf62d5dd7a39cfd46 Reviewed-on: https://webrtc-review.googlesource.com/c/106022 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25187}
2018-10-15 17:15:12 +02:00
"../rtc_base/system:rtc_export",
Remove asyncinvoker from PeerConnection. The callback that the asyncinvoker was being used for, will now use a safety flag to check if call_ is valid before issuing calls. Using the flag is a step towards removing the call_ptr_ variable but in this CL we're just looking at replacing use of the async invoker. The safety flag is cleared at the same time as call_ is, which prevents pending callbacks for that call instance from running. Also adding TODOs related to this change that will be followed upon in other CLs. Bug: webrtc:11988, webrtc:11992, webrtc:11993 Change-Id: If3986758af6d01d39b2db0cce82e57fc48be9d7f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185508 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32208}
2020-09-28 10:39:31 +02:00
"../rtc_base/task_utils:pending_task_safety_flag",
Remove use of AsyncInvoker from SctpDataChannel Bug: webrtc:11908 Change-Id: I4a32d9c3903d56994f2be365ab3fee2599ed6cdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184604 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32145}
2020-09-20 23:56:24 +02:00
"../rtc_base/task_utils:to_queued_task",
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",
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",
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
"../stats",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../system_wrappers",
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:field_trial",
"../system_wrappers:metrics",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42 Original change's description: > Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase > > Bug: None > Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad > Reviewed-on: https://webrtc-review.googlesource.com/c/107303 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25312} Tbr: kwiberg@webrtc.org Bug: None Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc Reviewed-on: https://webrtc-review.googlesource.com/c/107716 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-23 10:07:25 +02:00
"//third_party/abseil-cpp/absl/strings",
Replace rtc::Optional with absl::optional in pc This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameter 'pc' find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12 Reviewed-on: https://webrtc-review.googlesource.com/84181 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 16:47:43 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
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
}
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") {
sources = [
"connection_context.cc",
"connection_context.h",
]
deps = [
":rtc_pc_base",
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:callfactory_api",
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",
"../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",
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",
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/transport:field_trial_based_config",
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/transport:sctp_transport_factory_interface",
"../api/transport:webrtc_key_value_config",
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
"../media:rtc_data",
"../media:rtc_media_base",
"../p2p:rtc_p2p",
"../rtc_base",
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",
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",
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
]
}
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") {
sources = [
"peer_connection_message_handler.cc",
"peer_connection_message_handler.h",
]
deps = [
":stats_collector_interface",
"../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",
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_base",
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",
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",
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") {
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") {
sources = [
"rtp_transceiver.cc",
"rtp_transceiver.h",
]
deps = [
":rtc_pc_base",
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_sender",
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",
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",
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",
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",
"../media:rtc_media_base",
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:refcount",
"../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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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") {
sources = [
"rtp_transmission_manager.cc",
"rtp_transmission_manager.h",
]
deps = [
":audio_rtp_receiver",
":rtc_pc_base",
":rtp_receiver",
":rtp_sender",
":rtp_transceiver",
":stats_collector_interface",
":transceiver_list",
":usage_pattern",
":video_rtp_receiver",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
"../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",
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
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
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_base/third_party/sigslot",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("transceiver_list") {
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",
"../api:scoped_refptr",
"../rtc_base:checks",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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") {
sources = [
"rtp_receiver.cc",
"rtp_receiver.h",
]
deps = [
":media_stream",
":video_track_source",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/video:video_frame",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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") {
sources = [
"audio_rtp_receiver.cc",
"audio_rtp_receiver.h",
]
deps = [
":audio_track",
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
":media_stream",
":remote_audio_source",
":rtp_receiver",
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",
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",
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
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:refcount",
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 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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("video_rtp_receiver") {
sources = [
"video_rtp_receiver.cc",
"video_rtp_receiver.h",
]
deps = [
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
":jitter_buffer_delay_proxy",
":media_stream",
":rtp_receiver",
":video_rtp_track_source",
":video_track",
"../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",
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
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 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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("video_rtp_track_source") {
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",
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
"../media:rtc_media_base",
"../rtc_base",
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",
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") {
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",
"../rtc_base:refcount",
]
}
rtc_library("video_track") {
sources = [
"video_track.cc",
"video_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
"../api/video:video_frame",
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:rtc_base_approved",
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 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_source_set("jitter_buffer_delay_interface") {
sources = [ "jitter_buffer_delay_interface.h" ]
deps = [
"../media:rtc_media_base",
"../rtc_base:refcount",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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") {
sources = [ "sdp_state_provider.h" ]
deps = [
":rtc_pc_base",
"../api:libjingle_peerconnection_api",
]
}
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_source_set("jitter_buffer_delay_proxy") {
sources = [ "jitter_buffer_delay_proxy.h" ]
deps = [
":jitter_buffer_delay_interface",
"../api:libjingle_peerconnection_api",
"../media:rtc_media_base",
]
}
rtc_library("jitter_buffer_delay") {
sources = [
"jitter_buffer_delay.cc",
"jitter_buffer_delay.h",
]
deps = [
":jitter_buffer_delay_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",
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
"../media:rtc_media_base",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:safe_minmax",
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 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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("remote_audio_source") {
sources = [
"remote_audio_source.cc",
"remote_audio_source.h",
]
deps = [
":rtc_pc_base",
"../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",
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
"../media:rtc_media_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",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_base_approved",
"../rtc_base:safe_conversions",
"../rtc_base:stringutils",
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 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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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") {
sources = [
"rtp_sender.cc",
"rtp_sender.h",
]
deps = [
":dtmf_sender",
":stats_collector_interface",
"../api:audio_options_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: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",
"../api:scoped_refptr",
"../api/crypto:frame_encryptor_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
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base",
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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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") {
sources = [
"rtp_parameters_conversion.cc",
"rtp_parameters_conversion.h",
]
deps = [
":rtc_pc_base",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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") {
sources = [
"dtmf_sender.cc",
"dtmf_sender.h",
]
deps = [
"../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",
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:rtc_base",
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/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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_library("media_stream") {
sources = [
"media_stream.cc",
"media_stream.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:rtc_base",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../media:rtc_media_base",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base/system:rtc_export",
]
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
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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_source_set("stats_collector_interface") {
sources = [ "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
}
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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
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 = [
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
":peerconnection",
"../api:libjingle_peerconnection_api",
]
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 = [
(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
"channel_manager_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",
(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_filter_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
]
include_dirs = [ "//third_party/libsrtp/srtp" ]
if (is_win) {
libs = [ "strmiids.lib" ]
}
deps = [
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",
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",
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
":peerconnection",
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",
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
":rtc_pc_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
":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",
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",
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_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",
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",
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",
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",
SCTP: Treat message size zero as "responder selects" This also refactors some of the code in peerconnection for handling SCTP transports to be internal to the webrtc::SctpTransport class, rather than being in peerconnection. Bug: webrtc:10358, webrtc:10629 Change-Id: I15ecf95c199f56b08909e5a9311d446a412ed162 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137041 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27960}
2019-05-16 11:49:17 +02:00
"../media:rtc_data",
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
"../media:rtc_media_base",
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",
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",
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",
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",
"../p2p:rtc_p2p",
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",
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",
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 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_approved",
"../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:threading",
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",
guard flexfec usage with fieldtrial otherwise if the client receives a flexfec-enabled offer and receiving flexfec is enabled by default, an answer or subsequent offer will enable sending flexfec. BUG=webrtc:8151 Change-Id: I632094f69ffa68518b6b8f31175eb093efaf51c6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193862 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32628}
2020-11-17 21:59:12 +01:00
"../test:field_trial",
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",
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",
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) {
deps += [ "//third_party/libsrtp" ]
}
if (is_android) {
deps += [ "//testing/android/native_test:native_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
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",
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",
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",
[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",
"../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",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
[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",
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",
[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
"../modules/audio_device:audio_device_api",
"../modules/audio_processing: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
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../pc:peerconnection",
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",
[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",
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",
"../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:perf_test",
"../test:test_support",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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
}
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",
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",
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
"../pc:peerconnection",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:rtc_base_approved",
"../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("pc_test_utils") {
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
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/fake_audio_capture_module.cc",
"test/fake_audio_capture_module.h",
"test/fake_data_channel_provider.h",
"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",
Propagate base minimum delay from video jitter buffer to webrtc/api. On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface, GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes the same concept. Bug: webrtc:10287 Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957 Reviewed-on: https://webrtc-review.googlesource.com/c/123482 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26877}
2019-02-27 15:32:48 +01:00
"test/mock_delayable.h",
(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/mock_peer_connection_observers.h",
"test/mock_rtp_receiver_internal.h",
"test/mock_rtp_sender_internal.h",
"test/peer_connection_test_wrapper.cc",
"test/peer_connection_test_wrapper.h",
"test/rtc_stats_obtainer.h",
"test/test_sdp_strings.h",
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
]
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
":jitter_buffer_delay",
":jitter_buffer_delay_interface",
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
":libjingle_peerconnection",
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
":peerconnection",
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
":rtc_pc_base",
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",
":video_track_source",
[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",
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class Bug: webrtc:10138 Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-05 15:59:00 +01:00
"../api:create_frame_generator",
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",
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",
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",
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
"../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",
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",
[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",
"../api/audio_codecs:audio_codecs_api",
Use explicit TaskQueueFactory for FrameGeneratorCapturer in FrameGeneratorCapturerVideoTrackSource This replaces the implicit usage of GlobalTaskQueueFactory with an explicitly provided DefaultTaskQueueFactory instance. Bug: webrtc:10284 Change-Id: I40cbaa16181ab4b5a3528871cb068b09fe06b599 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133574 Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27695}
2019-04-18 14:02:11 +02:00
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
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",
Move VideoStreamEncoderInterface to api/. Bug: webrtc:8830 Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0 Reviewed-on: https://webrtc-review.googlesource.com/74481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 14:09:31 +02:00
"../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../api/video:video_rtp_headers",
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",
[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",
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
"../call:call_interfaces",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../media:rtc_data",
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
"../media:rtc_media",
Fix some dependencies for peerconnection_and_implicit_call_api. Some targets depend on "api/peerconnectioninterface.h" which is part of //api:peerconnection_and_implicit_call_api. Furthermore, peerconnection_and_implicit_call_api depends among other things of headers in //media:rtc_media_base, so we should add it as a dependency as well. Bug: webrtc:7504 Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac Reviewed-on: https://webrtc-review.googlesource.com/24660 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20815}
2017-11-20 18:41:50 +01:00
"../media:rtc_media_base",
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
"../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_device",
"../modules/audio_processing",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
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",
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",
[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
"../p2p:rtc_p2p",
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",
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",
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 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_approved",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
Use TaskQueueForTest instead or TaskQueue in unittests To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue Bug: webrtc:10284 Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:37:01 +01:00
"../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",
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",
Adds repeating task class. This CL adds a single class to manage the use case of having a task that repeats itself by a fixed or variable interval. It replaces the repeating task previously locally defined for rtp transport controller send as well as the cancelable periodic task. Furthermore, it is introduced where one off repeating tasks were created before. It provides the currently used functionality of the cancelable periodic task, but not some of the unused features, such as allowing cancellation of tasks before they are started and cancellation of a task after the owning task queue has been destroyed. Bug: webrtc:9883 Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca Reviewed-on: https://webrtc-review.googlesource.com/c/116182 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26313}
2019-01-18 10:30:54 +01:00
"../rtc_base/task_utils:repeating_task",
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",
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
"../test:test_support",
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:video_test_common",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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
}
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 = [
(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",
"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",
"peer_connection_end_to_end_unittest.cc",
"peer_connection_factory_unittest.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",
(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",
"sdp_serializer_unittest.cc",
"stats_collector_unittest.cc",
"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
]
if (rtc_enable_sctp) {
Prefix HAVE_SCTP macro with WEBRTC_. Generated automatically with: git grep -l "\bHAVE_SCTP\b" | xargs \ sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g' Bug: webrtc:11142 Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33042}
2021-01-18 13:24:40 +01:00
defines = [ "WEBRTC_HAVE_SCTP" ]
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 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",
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",
":jitter_buffer_delay_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
":media_stream",
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
":peerconnection",
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",
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
":rtc_pc_base",
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 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",
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",
[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",
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",
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",
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",
[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",
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",
Add missing mock headers to api. R=mbonadei@webrtc.org Bug: webrtc:7618 Change-Id: Ia622a7623b2fa05ec14b52d5d31d158d1bd0ef6d Reviewed-on: https://webrtc-review.googlesource.com/34644 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21364}
2017-12-19 10:32:11 +01:00
"../api:mock_rtp",
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",
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",
[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",
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",
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",
"../api/task_queue:default_task_queue_factory",
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",
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",
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",
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",
[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",
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",
[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",
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
"../media:rtc_media_engine_defaults",
[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
"../modules/audio_device:audio_device_api",
"../modules/audio_processing:audio_processing_statistics",
APM: Add build flag to allow building WebRTC without APM This CL adds a build flag to allow building the non-test parts of WebRTC without the audio processing module. The CL also ensures that the WebRTC code correctly handles the case when no APM is available. Bug: webrtc:5298 Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31133}
2020-04-26 23:56:17 +02:00
"../modules/audio_processing:audioproc_test_utils",
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",
Make ICE transports injectable. Bug: chromium:1024965 Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820 Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Honghai Zhang <honghaiz@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29807}
2019-11-15 12:33:05 -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",
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",
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",
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",
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",
"../rtc_base:threading",
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",
Reland "Surface ICE candidates that match an updated candidate filter." This is a reland of cd8d1cf68e4eeed71fba51c97006a91bfd41813d Original change's description: > Surface ICE candidates that match an updated candidate filter. > > After this change an ICE agent can surface candidates that do not match > the previous filter but are allowed by the updated one. The candidate > filter, as part of the internal implementation in the ICE transport, > manifests the RTCIceTransportPolicy field in RTCConfiguration. > > This new feature would allow an ICE agent to gather new candidates when > the transport policy changes from e.g. 'relay' to 'all' without an ICE > restart. > > A caveat in the current implementation remains, and a candidate can > surface multiple times if the transport policy, or the candidate filter > directly, performs multiple transitions from a value that disallows to > one that allows the underlying candidate type. For example, if the > transport policy is updated by 'all' -> 'relay' -> 'all', the same host > candidate can surface after the second update. > > > Bug: webrtc:8939 > Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282 > Commit-Queue: Qingsi Wang <qingsi@webrtc.org> > Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org> > Reviewed-by: Seth Hampson <shampson@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27674} Bug: webrtc:8939 Change-Id: I9c32b1ea05028ecd937ab4912779dd958faf734f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133582 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org> Commit-Queue: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27694}
2019-04-18 10:41:58 -07:00
"../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../test:fileutils",
Make the RtpHeaderParserImpl available to tests and tools only. There are a few reasons for making this test only: * The code is only used by tests and utilities. * The pure interface has only a single implementation so an interface isn't really needed. (a followup change could remove it altogether) * The implementation always incorporates locking regardless of how the class gets used. See e.g. previous use in the Packet class. * The implementation is a layer on top of RtpUtility::RtpHeaderParser which is sufficient for most production cases. Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36 Bug: none Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 16:39:05 +02:00
"../test:rtp_test_utils",
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",
Adds field trial to separate audio and video packets for delay-based overuse detection. The decision to route audio packets to a separate overuse detector is off by default and requires the field trial WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/ The parameters control the threshold for switching over to the audio overuse detector if we stop receiving feedback for video. Bug: webrtc:10932 Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30694}
2020-03-05 16:52:10 +01:00
"./scenario_tests:pc_scenario_tests",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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
]
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) {
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 += [ ":android_black_magic" ]
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",
Rename Call::Config to CallConfig, keep old name as alias. We want api/peerconnectioninterface.h (and corresponding build target) to not depend on call.h, and generally we treat Call as an internal, non-api, class. But we need CallFactoryInterface in the api in order to enable use of PeerConnection with or without support for media. Making CallConfig a top-level class makes it possible to forward declare it, together with Call, for use in callfactoryinterface.h and peerconnectioninterface.h. Delete the peerconnection_and_implicit_call_api target, replaced by new target callfactory_api, to link between Call and Peerconnection. Bug: webrtc:7504 Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1 Reviewed-on: https://webrtc-review.googlesource.com/46201 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22020}
2018-02-14 12:20:13 +01:00
"../api:callfactory_api",
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",
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",
Insert frame transformer between Encoded and Packetizer. Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27 16:16:55 +01:00
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp
# constant.
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_media_base",
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",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
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/utility",
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",
"../p2p:rtc_p2p",
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
"../pc:rtc_pc",
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",
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
"../rtc_base:rtc_base_approved",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
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
]
if (is_android) {
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 += [
"//testing/android/native_test:native_test_support",
# We need to depend on this one directly, or classloads will fail for
# the voice engine BuildInfo, for instance.
"../sdk/android:libjingle_peerconnection_java",
]
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
shard_timeout = 900
}
}
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") {
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
# The android code uses hacky includes to chromium-base and the ssl code;
# having this in a separate target enables us to keep the peerconnection
# unit tests clean.
check_includes = false
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 = [
"../sdk/android:libjingle_peerconnection_jni",
"//testing/android/native_test:native_test_support",
]
}
}
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
}
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