admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/media/BUILD.gn

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

1101 lines
34 KiB
Plaintext
Raw Normal View History

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.
import("//build/config/linux/pkg_config.gni")
Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." This reverts commit f8c81ca4690aa3e470cc61633f512de383c308a8. Reason for revert: <Prepare to reland> Original change's description: > Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." > > This reverts commit 256733c78af655029cb04afae2c404afb41ea685. > > Reason for revert: <breaks downstream> > > Original change's description: > > Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate. > > > > Bug: webrtc:13573 > > Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36940} > > Bug: webrtc:13573 > Change-Id: I3341b6b96a56de63058c38943611b8c1629294ce > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262941 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36942} Bug: webrtc:13573 Change-Id: Iaf1222c58a18f378df20e4f83262b9a9da491712 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262943 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36985}
2022-05-24 10:32:32 +02:00
import("//third_party/libaom/options.gni")
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")
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("media") {
Update gn files to support Mozilla build Bug: webrtc:8670 No-Presubmit: true Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa Reviewed-on: https://webrtc-review.googlesource.com/38920 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-15 10:20:00 -05:00
deps = []
if (!build_with_mozilla) {
deps += [
":rtc_media",
":rtc_media_base",
]
}
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_media_defines_config") {
Remove HAVE_WEBRTC_VOICE. Appears not used anymore. Bug: none Change-Id: Ic2238e6ad3d9917208bdb4a101f1ce254b1272ac Reviewed-on: https://webrtc-review.googlesource.com/c/120963 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26578}
2019-02-05 12:43:03 +01:00
defines = [ "HAVE_WEBRTC_VIDEO" ]
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
}
Move MediaConfig to its own header file and target. To eliminate circular dependencies, we need to eliminate the include of media/base/mediachannel.h from api/peerconnectioninterface.h. MediaConfig is one of the types the PeerConnection api depends on, since it's part of PeerConnectionInterface::RTCConfiguration. It's formally a public member, but the intention is that applications should use accessor mehtods on RTCConfiguration and never access the contents of MediaConfig directly. Bug: webrtc:7504 Change-Id: Idfab6f69132d6b90d1628fa4543a393e22db79ac Reviewed-on: https://webrtc-review.googlesource.com/41260 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21731}
2018-01-23 10:37:42 +01:00
rtc_source_set("rtc_media_config") {
visibility = [ "*" ]
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "base/media_config.h" ]
Move MediaConfig to its own header file and target. To eliminate circular dependencies, we need to eliminate the include of media/base/mediachannel.h from api/peerconnectioninterface.h. MediaConfig is one of the types the PeerConnection api depends on, since it's part of PeerConnectionInterface::RTCConfiguration. It's formally a public member, but the intention is that applications should use accessor mehtods on RTCConfiguration and never access the contents of MediaConfig directly. Bug: webrtc:7504 Change-Id: Idfab6f69132d6b90d1628fa4543a393e22db79ac Reviewed-on: https://webrtc-review.googlesource.com/41260 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21731}
2018-01-23 10:37:42 +01:00
}
Move h264_profile_level_id and vp9_profile to api/video_codecs This is a refactor to simplify a follow-up CL of adding SdpVideoFormat::IsSameCodec. The original files media/base/h264_profile_level_id.* and media/base/vp9_profile.h must be kept until downstream projects stop using them. Bug: chroimium:1187565 Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-19 09:09:26 +02:00
rtc_library("rtc_sdp_video_format_utils") {
Parse max-fr and max-fs from SDP FMTP line max-fr and max-fs are mandatory fields for VP8 and VP9. Add parsing as a first step to enable use of these fields. Bug: chromium:1032518 Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30114}
2019-12-19 15:05:20 +01:00
visibility = [ "*" ]
sources = [
Move h264_profile_level_id and vp9_profile to api/video_codecs This is a refactor to simplify a follow-up CL of adding SdpVideoFormat::IsSameCodec. The original files media/base/h264_profile_level_id.* and media/base/vp9_profile.h must be kept until downstream projects stop using them. Bug: chroimium:1187565 Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-19 09:09:26 +02:00
"base/sdp_video_format_utils.cc",
"base/sdp_video_format_utils.h",
Parse max-fr and max-fs from SDP FMTP line max-fr and max-fs are mandatory fields for VP8 and VP9. Add parsing as a first step to enable use of these fields. Bug: chromium:1032518 Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30114}
2019-12-19 15:05:20 +01:00
]
deps = [
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs which needs to be added to the remote codecs a=fmtp: This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes. This parameter allows for large-scale experimentation and A/B testing whether the new behavior has advantages. It is to be considered transitional and may be removed again in the future. BUG=webrtc:10107 Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41805}
2024-02-20 15:28:14 +01:00
":media_constants",
Parse max-fr and max-fs from SDP FMTP line max-fr and max-fs are mandatory fields for VP8 and VP9. Add parsing as a first step to enable use of these fields. Bug: chromium:1032518 Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30114}
2019-12-19 15:05:20 +01:00
"../api/video_codecs:video_codecs_api",
Move h264_profile_level_id and vp9_profile to api/video_codecs This is a refactor to simplify a follow-up CL of adding SdpVideoFormat::IsSameCodec. The original files media/base/h264_profile_level_id.* and media/base/vp9_profile.h must be kept until downstream projects stop using them. Bug: chroimium:1187565 Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-19 09:09:26 +02:00
"../rtc_base:checks",
Parse max-fr and max-fs from SDP FMTP line max-fr and max-fs are mandatory fields for VP8 and VP9. Add parsing as a first step to enable use of these fields. Bug: chromium:1032518 Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30114}
2019-12-19 15:05:20 +01:00
"../rtc_base:stringutils",
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs which needs to be added to the remote codecs a=fmtp: This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes. This parameter allows for large-scale experimentation and A/B testing whether the new behavior has advantages. It is to be considered transitional and may be removed again in the future. BUG=webrtc:10107 Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41805}
2024-02-20 15:28:14 +01:00
"//third_party/abseil-cpp/absl/algorithm:container",
]
Parse max-fr and max-fs from SDP FMTP line max-fr and max-fs are mandatory fields for VP8 and VP9. Add parsing as a first step to enable use of these fields. Bug: chromium:1032518 Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30114}
2019-12-19 15:05:20 +01:00
}
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_media_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 = []
libs = []
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
sources = [
"base/adapted_video_track_source.h", # Used downstream
"base/video_adapter.h", # Used once downstream
"base/video_broadcaster.h", # Used downstream
"base/video_common.h", # Used downstream
]
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
deps = [
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":adapted_video_track_source",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":codec",
":media_channel",
":media_channel_impl",
":rid_description",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
":rtc_media_config",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":rtp_utils",
":stream_params",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":video_adapter",
":video_broadcaster",
":video_common",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":video_source_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
"../api:array_view",
Move AudioOptions to its own header file and target. It is part of our api. With the intention to later delete the inclusion of mediachannel.h from api/peerconnectioninterface.h, and eliminate circular dependencies. Bug: webrtc:7504 Change-Id: If44efd14d85675530e457760a1c4a1d338f931b7 Reviewed-on: https://webrtc-review.googlesource.com/41281 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21694}
2018-01-19 11:28:54 +01:00
"../api:audio_options_api",
Apply PIMPL pattern to MediaSender and Receiver objects This detaches the implementation (which is still merged) from the objects used to interface to it. Bug: webrtc:13931 Change-Id: I872ee10e4ed9fa432bfa231f723af1d3989d79d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288080 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38906}
2022-12-15 16:49:13 +00:00
"../api:call_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
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",
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",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
2022-10-18 17:05:16 +02:00
"../api:rtp_sender_interface",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
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",
Split audio and video channels into Send and Receive APIs. The implementation here has a number of changes that force the callers that called the "channel" functions into specific interfaces rather than just letting C++ take care of it; this should go away once there stops being a common implementation class for those interfaces. Bug: webrtc:13931 Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-13 12:57:24 +00:00
"../api:transport_api",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../api/audio:audio_frame_processor",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../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",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Split audio and video channels into Send and Receive APIs. The implementation here has a number of changes that force the callers that called the "channel" functions into specific interfaces rather than just letting C++ take care of it; this should go away once there stops being a common implementation class for those interfaces. Bug: webrtc:13931 Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-13 12:57:24 +00:00
"../api/task_queue",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
datachannel: Merge SendDataParams and DMT types with webrtc equivalent cricket::SendDataParams is replaced by webrtc::SendDataParams. cricket::DataMessageType is replaced by webrtc::DataMessageType. The sid member from cricket::SendDataParams is now passed as an argument to functions that used one when necessary. Bug: webrtc:7484 Change-Id: Ia4a89c9651fb54ab9a084a6098d49130b6319e1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217761 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33966}
2021-05-10 11:29:56 +02:00
"../api/transport:datagram_transport_interface",
Move stun.h to api/. We now have two downstream users of stun.h, so it appears to be generally usable. I put this in a new dir networking/, but I'm open to suggestions here (maybe some things in api/ should move in there). I checked what our downstream users are actually using, and it's cricket::ComputeStunCredentialHash cricket::<constants> cricket::TurnMessage cricket::GetStunErrorResponseType cricket::StunAttribute::CreateAddress cricket::StunErrorCodeAttribute cricket::StunByteStringAttribute StunAttribute::CreateUnknownAttributes cricket::TurnErrorType cricket::StunMessage I reckoned that was pretty much everything in stun.h, so I didn't bother splitting it up. They don't use every function and constant in there, but all _types_ of functions and constants, so for the sake of coherence I don't think it makes sense to split it. There's some old stuff in there like GTURN which could arguably be split out, but it should likely go away soon anyway, so I don't think it's worth the effort. Steps: 1) land this 2) update downstream to point to the new header and target 3) remove p2p/base:stun_types. Bug: webrtc:11091 Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29822}
2019-11-18 15:53:32 +01:00
"../api/transport:stun_types",
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",
Reland "Wire up non-sender RTT for audio, and implement related standardized stats." This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb. Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2. Original change's description: > Revert "Wire up non-sender RTT for audio, and implement related standardized stats." > > This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e. > > Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium. > > Original change's description: > > Wire up non-sender RTT for audio, and implement related standardized stats. > > > > The implemented stats are: > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime > > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements > > > > Bug: webrtc:12951, webrtc:12714 > > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956 > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34861} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta,hbos,minyue > > Bug: webrtc:12951, webrtc:12714 > Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Reviewed-by: Olga Sharonova <olka@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34897} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12951, webrtc:12714 Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34930}
2021-09-03 14:51:22 +00:00
"../api/units:time_delta",
Apply PIMPL pattern to MediaSender and Receiver objects This detaches the implementation (which is still merged) from the objects used to interface to it. Bug: webrtc:13931 Change-Id: I872ee10e4ed9fa432bfa231f723af1d3989d79d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288080 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38906}
2022-12-15 16:49:13 +00:00
"../api/video:recordable_encoded_frame",
VideoAdapter: Interpret requested resolution as max restriction. The `requested_resolution` API must not change aspect ratio, example: - Frame is 60x30 - Requested is 30x30 - We expect 30x15 (not 30x30!) as to maintain aspect ratio. This bug was previously fixed by making VideoAdapter unaware of the requested resolution behind a flag: this seemed OK since the VideoStreamEncoder ultimately decides the resolution, whether or not the incoming frame is adapted. But this is not desired for some non-Chrome use cases. This CL attempts to make both Chrome and non-Chrome use cases happy by implementing the aspect ratio preserving restriction inside VideoAdapter too. This allows us to get rid of the "use_standard_requested_resolution" flag and change the "VideoStreamEncoderResolutionTest" TEST_P to TEST_F. Bug: webrtc:366067962, webrtc:366284861 Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720 Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43037}
2024-09-17 15:13:36 +02:00
"../api/video:resolution",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video:video_bitrate_allocation",
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:video_bitrate_allocator_factory",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +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",
Add scalability mode to RTCOutboundRtpStreamStats stats This is in the webrtc-stats spec at https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode. This adds the scalability mode to CodecSpecificInfo which is used to plumb the modes for each simulcast layer. TBR=orphis@webrtc.org Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals. Bug: webrtc:14730 Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38847}
2022-12-06 10:09:10 +00:00
"../api/video_codecs:scalability_mode",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../common_video",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../modules/async_audio_processing",
Add RTCAudioPlayoutStats to GetStats(). This is done by allowing implementations of AudioDeviceModule to implement the GetStats() method. The default implementation returns nullopt, in which case RTCAudioPlayoutStats will not be visible in the stats. Bug: webrtc:14653 Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-13 16:42:36 +01:00
"../modules/audio_device",
[video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was split up into smaller pieces for reviewability. Spec: https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats In [1], ReportBlockData was implemented and tested. This CL adds the plumbing to make it available in MediaSenderInfo for video streams, but the code is not wired up to make use of these stats. In follow-up CL [2], ReportBlockData will be used to implement RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration tests exercising the full code path. [1] https://webrtc-review.googlesource.com/c/src/+/136584 [2] https://webrtc-review.googlesource.com/c/src/+/138067 Bug: webrtc:10456 Change-Id: Icd20452cb4b4908203b28ae9d9f52c25693cf91d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138065 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28071}
2019-05-27 10:44:24 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:dscp",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
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:sanitizer",
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",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Split audio and video channels into Send and Receive APIs. The implementation here has a number of changes that force the callers that called the "channel" functions into specific interfaces rather than just letting C++ take care of it; this should go away once there stops being a common implementation class for those interfaces. Bug: webrtc:13931 Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38888}
2022-12-13 12:57:24 +00:00
"../rtc_base/network:sent_packet",
Migrate media/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I69e4a1b37737ac8dd852a032612623c4c4f3a30b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176744 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31648}
2020-07-07 15:43:11 +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 "Fix race between enabled() and set_enabled() in VideoTrack." This reverts commit 096ad02c02b4bc6c046282b8793ef84d041dd0d8. Reason for revert: Including a fix for the test issue. Original change's description: > Revert "Fix race between enabled() and set_enabled() in VideoTrack." > > This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a. > > Reason for revert: Breaks Chromium Android browser tests on fyi bots. > > Original change's description: > > Fix race between enabled() and set_enabled() in VideoTrack. > > > > Along the way I introduced VideoSourceBaseGuarded, which is equivalent > > to VideoSourceBase except that it applies thread checks. I found that > > it's easy to use VideoSourceBase incorrectly and in fact there appear > > to be tests that do this. > > > > I made the source object const in VideoTrack, as it already was in > > AudioTrack, and that allowed for making the GetSource() accessors > > bypass the proxy thread hop and give the caller direct access. > > > > Bug: webrtc:12773, b/188139639, webrtc:12780 > > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > > Commit-Queue: Tommi <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34096} > > TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12773 > Bug: b/188139639 > Bug: webrtc:12780 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637 > Reviewed-by: Evan Shrubsole <eshr@google.com> > Commit-Queue: Evan Shrubsole <eshr@google.com> > Cr-Commit-Position: refs/heads/master@{#34101} # Not skipping CQ checks because this is a reland. Bug: webrtc:12773 Bug: b/188139639 Bug: webrtc:12780 Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34104}
2021-05-24 16:54:41 +02:00
"../rtc_base/system:no_unique_address",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
Move VideoEncoderConfig from api/ into video/config This cl move VideoEncoderConfig from api/ to video/config. VideoStreamEncoderInterface and VideoStreamEncoderObserver are moved as collateral. brandt@ think that the reason these were in api/ in the first place had to downstream project. Functionality wise, this is a NOP, but it makes it easier to modify the encoder (config). Bug: webrtc:14451 Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 10:28:24 +02:00
"../video/config:encoder_config",
Replace cricket::LeastCommonMultiple and cricket::GreatestCommonDivisor with std::lcm and std::gcd. The std::lcm and std::gcd functions are part of the C++ standard library. The existing functions are marked as deprecated rather than deleted in the case of possible third party uses. #rtc_cleanup Bug: webrtc:377205743 Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@google.com> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43368}
2024-11-06 15:14:40 +00:00
"//third_party/abseil-cpp/absl/base:core_headers",
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
]
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
}
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
rtc_library("adapted_video_track_source") {
sources = [
"base/adapted_video_track_source.cc",
"base/adapted_video_track_source.h",
]
deps = [
":video_adapter",
":video_broadcaster",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/system:rtc_export",
]
}
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
rtc_source_set("audio_source") {
sources = [ "base/audio_source.h" ]
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
}
rtc_library("video_adapter") {
sources = [
"base/video_adapter.cc",
"base/video_adapter.h",
]
deps = [
":video_common",
VideoAdapter: Interpret requested resolution as max restriction. The `requested_resolution` API must not change aspect ratio, example: - Frame is 60x30 - Requested is 30x30 - We expect 30x15 (not 30x30!) as to maintain aspect ratio. This bug was previously fixed by making VideoAdapter unaware of the requested resolution behind a flag: this seemed OK since the VideoStreamEncoder ultimately decides the resolution, whether or not the incoming frame is adapted. But this is not desired for some non-Chrome use cases. This CL attempts to make both Chrome and non-Chrome use cases happy by implementing the aspect ratio preserving restriction inside VideoAdapter too. This allows us to get rid of the "use_standard_requested_resolution" flag and change the "VideoStreamEncoderResolutionTest" TEST_P to TEST_F. Bug: webrtc:366067962, webrtc:366284861 Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720 Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43037}
2024-09-17 15:13:36 +02:00
"../api/video:resolution",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
"../api/video:video_frame",
"../common_video",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/system:rtc_export",
"../system_wrappers:field_trial",
]
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
}
rtc_library("video_source_base") {
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
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
"base/video_source_base.cc",
"base/video_source_base.h",
Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ ) Reason for revert: Include fix for downstream import. Original issue's description: > Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ ) > > Reason for revert: > Breaks downstream import. > > Original issue's description: > > Split out target rtc_media_base from rtc_media > > > > The purpose with this CL is to be able to depend on > > cricket::VideoCodec (webrtc/media/base/codec.h) from other targets > > without getting cyclic dependencies. > > > > BUG=webrtc:6402,webrtc:6337 > > > > NOTRY=True > > > > Committed: https://crrev.com/aae7e7cf35a5bb43ebbaf75396aa7ccc544e920a > > Cr-Commit-Position: refs/heads/master@{#15137} > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6402,webrtc:6337 > > Committed: https://crrev.com/0d0d7531b50a78efe7468610395e9dc5f496e2e9 > Cr-Commit-Position: refs/heads/master@{#15139} TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6402,webrtc:6337 Review-Url: https://codereview.webrtc.org/2509123003 Cr-Commit-Position: refs/heads/master@{#15235}
2016-11-25 00:40:18 -08:00
]
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
deps = [
"../api:sequence_checker",
"../api/video:video_frame",
"../rtc_base:checks",
"../rtc_base/system:no_unique_address",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
]
}
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
rtc_library("video_broadcaster") {
sources = [
"base/video_broadcaster.cc",
"base/video_broadcaster.h", # Used downstream
]
deps = [
":video_common",
":video_source_base",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
Replace cricket::LeastCommonMultiple and cricket::GreatestCommonDivisor with std::lcm and std::gcd. The std::lcm and std::gcd functions are part of the C++ standard library. The existing functions are marked as deprecated rather than deleted in the case of possible third party uses. #rtc_cleanup Bug: webrtc:377205743 Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@google.com> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43368}
2024-11-06 15:14:40 +00:00
"../api:video_track_source_constraints",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base/synchronization:mutex",
]
}
rtc_library("video_common") {
sources = [
"base/video_common.cc",
"base/video_common.h",
]
deps = [
"../api:array_view",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../rtc_base/system:rtc_export",
Replace cricket::LeastCommonMultiple and cricket::GreatestCommonDivisor with std::lcm and std::gcd. The std::lcm and std::gcd functions are part of the C++ standard library. The existing functions are marked as deprecated rather than deleted in the case of possible third party uses. #rtc_cleanup Bug: webrtc:377205743 Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Evan Shrubsole <eshr@google.com> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43368}
2024-11-06 15:14:40 +00:00
"//third_party/abseil-cpp/absl/base:core_headers",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
]
}
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
rtc_library("media_engine") {
sources = [
"base/media_engine.cc",
"base/media_engine.h",
]
deps = [
":codec",
":media_channel",
":media_channel_impl",
":rtc_media_config",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
":stream_params",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":video_common",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../api:array_view",
"../api:audio_options_api",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../api:field_trials_view",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../api:rtc_error",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../api:rtp_parameters",
Move the payload type picker to call/ Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00
"../api:scoped_refptr",
"../api/audio:audio_device",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:options",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
"../call:call_interfaces",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../rtc_base/system:file_wrapper",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
]
Reland of Split out target rtc_media_base from rtc_media (patchset #1 id:1 of https://codereview.webrtc.org/2508163002/ ) Reason for revert: Include fix for downstream import. Original issue's description: > Revert of Split out target rtc_media_base from rtc_media (patchset #3 id:40001 of https://codereview.webrtc.org/2471573003/ ) > > Reason for revert: > Breaks downstream import. > > Original issue's description: > > Split out target rtc_media_base from rtc_media > > > > The purpose with this CL is to be able to depend on > > cricket::VideoCodec (webrtc/media/base/codec.h) from other targets > > without getting cyclic dependencies. > > > > BUG=webrtc:6402,webrtc:6337 > > > > NOTRY=True > > > > Committed: https://crrev.com/aae7e7cf35a5bb43ebbaf75396aa7ccc544e920a > > Cr-Commit-Position: refs/heads/master@{#15137} > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6402,webrtc:6337 > > Committed: https://crrev.com/0d0d7531b50a78efe7468610395e9dc5f496e2e9 > Cr-Commit-Position: refs/heads/master@{#15139} TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6402,webrtc:6337 Review-Url: https://codereview.webrtc.org/2509123003 Cr-Commit-Position: refs/heads/master@{#15235}
2016-11-25 00:40:18 -08:00
}
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
rtc_library("media_channel_impl") {
sources = [
"base/media_channel_impl.cc",
"base/media_channel_impl.h",
]
deps = [
":codec",
":media_channel",
":rtp_utils",
":stream_params",
"../api:audio_options_api",
"../api:call_api",
"../api:frame_transformer_interface",
"../api:media_stream_interface",
"../api:rtc_error",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:rtp_headers",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:scalability_mode",
"../api/video_codecs:video_codecs_api",
"../common_video",
"../modules/rtp_rtcp:rtp_rtcp_format",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:dscp",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:logging",
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:socket",
"../rtc_base/network:sent_packet",
"//third_party/abseil-cpp/absl/functional:any_invocable",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
]
}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
deps = [
":codec",
":media_constants",
":rtp_utils",
":stream_params",
"../api:audio_options_api",
"../api:call_api",
"../api:frame_transformer_interface",
"../api:media_stream_interface",
"../api:rtc_error",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
"../api:rtp_headers",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../api:rtp_parameters",
"../api:rtp_sender_interface",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
"../api:scoped_refptr",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing_statistics",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:datagram_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
Propagate time of the last received packet with Timestamp type Bug: webrtc:13757 Change-Id: I446fc10ad6a90ab9ecaac337b9f2ad4ccad37cbd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307020 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40211}
2023-06-02 14:54:45 +02:00
"../api/units:timestamp",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
"../api/video:recordable_encoded_frame",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:scalability_mode",
"../api/video_codecs:video_codecs_api",
"../common_video",
"../modules/rtp_rtcp:rtp_rtcp_format",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:buffer",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
"../rtc_base:checks",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:copy_on_write_buffer",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:dscp",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:logging",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:socket",
"../rtc_base:stringutils",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
"../rtc_base/network:sent_packet",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../video/config:encoder_config",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
Make a shim object implementing the VideoMediaChannel interface The intent is that this object can be used instead of VideoMediaChannel, clearing the way for decomposing VideoMediaChannel into send and receive classes. This CL uses it for the "both" role of WebRtcVideoEngine::CreateMediaChannel; a later CL will use it for all roles on all engines. Bug: webrtc:13931 Change-Id: Ibd0ca2c3c45b5e3bfcced8f7e30a1edd63cf7654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306720 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40173}
2023-05-30 04:04:07 +00:00
]
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
}
rtc_library("codec") {
sources = [
"base/codec.cc",
"base/codec.h",
Move some codec-comparing functions to a single file. This CL is a pure move; later CLs will try to increase consistency between the functions. Bug: webrtc:360058654 Change-Id: I6662b3d35f8e2dab60c2778a4755454fe3029fe2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43210}
2024-10-09 14:05:51 +00:00
# Because Codec::Matches uses a function from codec_comparators,
# there's a mutual dependency between these two files.
"base/codec_comparators.cc",
"base/codec_comparators.h",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
]
deps = [
":media_constants",
"../api:rtp_parameters",
Move GetSendCodec() to MediaSendChannelInterface This allows the voice send channels to share the method definition. Bug: webrtc:15214 Change-Id: Ie0cc23f3694eeb8322a9ea7328a8d56fa7571c95 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40322}
2023-06-20 17:12:57 +00:00
"../api/audio_codecs:audio_codecs_api",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api/video_codecs:scalability_mode",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../api/video_codecs:video_codecs_api",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
Refactor FindMatchingCodec This is in preparation for making a matcher that checks the parameters when all payload types come from the same number space. Bug: webrtc:360058654 Change-Id: Ibcf4fee8d882eb0fa7f83faf0278bc6757761e18 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365361 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43223}
2024-10-11 09:59:01 +00:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"//third_party/abseil-cpp/absl/strings",
Add AbslStringify for cricket::Codec This makes debug output easier to read. Bug: webrtc:360058654 Change-Id: I887be638489cde26868db0db2950262255213160 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365144 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43216}
2024-10-10 12:31:47 +00:00
"//third_party/abseil-cpp/absl/strings:str_format",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
]
}
rtc_library("rtp_utils") {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
]
deps = [
":turn_utils",
"../api:array_view",
"../modules/rtp_rtcp:rtp_rtcp_format",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base:byte_order",
"../rtc_base:checks",
Reland "Split digest methods from ssl target into digest target" This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4 Original change's description: > Split digest methods from ssl target into digest target > > in an attempt to break up the monolithic ssl target. > > BUG=None > > Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@meta.com> > Cr-Commit-Position: refs/heads/main@{#42249} Bug: webrtc:339300437 Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42304}
2024-05-14 08:35:24 -07:00
"../rtc_base:digest",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
"../rtc_base/system:rtc_export",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
]
}
rtc_library("stream_params") {
sources = [
"base/stream_params.cc",
"base/stream_params.h",
]
deps = [
":rid_description",
"../api:array_view",
"../rtc_base:stringutils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:unique_id_generator",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
]
}
rtc_library("media_constants") {
sources = [
"base/media_constants.cc",
"base/media_constants.h",
]
deps = [ "../rtc_base/system:rtc_export" ]
}
rtc_library("turn_utils") {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
]
deps = [
"../api/transport:stun_types",
"../rtc_base:byte_order",
"../rtc_base/system:rtc_export",
]
}
rtc_library("rid_description") {
sources = [
"base/rid_description.cc",
"base/rid_description.h",
]
deps = []
}
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_simulcast_encoder_adapter") {
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
visibility = [ "*" ]
defines = []
libs = []
sources = [
"engine/simulcast_encoder_adapter.cc",
"engine/simulcast_encoder_adapter.h",
]
deps = [
Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs which needs to be added to the remote codecs a=fmtp: This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes. This parameter allows for large-scale experimentation and A/B testing whether the new behavior has advantages. It is to be considered transitional and may be removed again in the future. BUG=webrtc:10107 Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41805}
2024-02-20 15:28:14 +01:00
":rtc_sdp_video_format_utils",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":video_common",
Fix spatial layers to 1 when doing VP9 simulcast encoding Libvpx works without this, so the existing tests pass. However, other encoder implementations (like rtc_video_encoder in Chrome) look at different fields and get confused about the configuration. Test: Integration tests with Chrome and windows hardware encoders. Bug: webrtc:348342168 Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42555}
2024-06-25 13:54:45 +00:00
"../api:array_view",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 19/inf Convert most field trials used in PCLF tests. Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d Bug: webrtc:10335 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703 Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40909}
2023-10-10 15:21:29 +02:00
"../api:field_trials_view",
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",
Propagate Environment through SimulcastEncoderAdapter when provided Bug: webrtc:15860 Change-Id: Iabd7752ada2f8f774de1e2adc02a4157004bf43c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342720 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41893}
2024-03-12 16:58:18 +01:00
"../api/environment",
Fix spatial layers to 1 when doing VP9 simulcast encoding Libvpx works without this, so the existing tests pass. However, other encoder implementations (like rtc_video_encoder in Chrome) look at different fields and get confused about the configuration. Test: Integration tests with Chrome and windows hardware encoders. Bug: webrtc:348342168 Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42555}
2024-06-25 13:54:45 +00:00
"../api/units:data_rate",
"../api/units:timestamp",
"../api/video:encoded_image",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator",
Move some video codec constants to separate file. kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't really beling on VideoBitrateAllocation. common_types.h is going away and it feels dubious to requrie include of the full VideoEncoder api to use them. Therefore moving them into a seprate file/target. Also includes some remaining cleanup of includes. Bug: webrtc:9271 Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697 Reviewed-on: https://webrtc-review.googlesource.com/c/117305 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26299}
2019-01-16 17:10:57 +01:00
"../api/video:video_codec_constants",
[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:video_frame",
Fix spatial layers to 1 when doing VP9 simulcast encoding Libvpx works without this, so the existing tests pass. However, other encoder implementations (like rtc_video_encoder in Chrome) look at different fields and get confused about the configuration. Test: Integration tests with Chrome and windows hardware encoders. Bug: webrtc:348342168 Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42555}
2024-06-25 13:54:45 +00:00
"../api/video:video_frame_type",
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",
SimulcastEncoderAdapter: support per layer fallback and single encoder proxying This CL adds an optional second encoder factory to SimulcastEncoderAdapter, that can be used to create software fallback adapter per simulcast layer. It also adds logic to check if the encoder supports simulcast natively, if so it only allocates a single instance and delegates the simulcast logic to that encoder instead. This means we will be able to remove EncoderSimulcastProxy. Bug: webrtc:11000 Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29364}
2019-10-01 18:50:03 +02:00
"../api/video_codecs:rtc_software_fallback_wrappers",
Fix spatial layers to 1 when doing VP9 simulcast encoding Libvpx works without this, so the existing tests pass. However, other encoder implementations (like rtc_video_encoder in Chrome) look at different fields and get confused about the configuration. Test: Integration tests with Chrome and windows hardware encoders. Bug: webrtc:348342168 Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42555}
2024-06-25 13:54:45 +00:00
"../api/video_codecs:scalability_mode",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
"../api/video_codecs:video_codecs_api",
SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. Results from test (CallPerfTest.TestEncodeFramerateVp8Simulcast): Simulcast streams: 0: max_fps:20 -> StreamStats.encode_frame_rate:15 (before), 20 (after) 1: max_fps:30 Bug: webrtc:13031 Change-Id: I30e6b2dcb2746859bd3e21b098bfa7b0fb3b2dda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230120 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34867}
2021-08-26 12:04:27 +02:00
"../common_video",
Define default max Qp in media/base/media_constants kDefaultQpMax=56 was defined in multiple places. Move it to media_constants and split it into two: VPx/AV1 and H26x values. H26x value is set to 51 which is the max bitstream QP value for H264/5. This CL is expected to be a no-op because: 1. VideoCodec::qpMax value has not changed for VP8/9 and AV1. 2. VideoCodec::qpMax is currently not used by OpenH264 wrapper (wiring it up is out-of-scope of this CL). 3. Previous default qpMax=56 exceeded the max value for H26x (=51). External HW H26x encoders likely clamped it and used 51. Bug: webrtc:14852 Change-Id: I1d795e695dac5c78e86ed829b24281e61066f668 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324282 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40997}
2023-10-23 17:11:21 +02:00
"../media:media_constants",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
SEA discards inactive encoders in implementation name Inactive encoders are included in the string when they are paused due to bitrate allocation being 0 for that simulcast layer. #rtc_ktlo Bug: webrtc:376804631 Change-Id: I4234b452b60fee58981907380df41962fda5bf40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43367}
2024-11-06 13:33:21 +00:00
"../rtc_base:stringutils",
Add resolution_bitrate_limits to EncoderInfo field trial. Added class EncoderInfoSettings for parsing settings. Added use of class to SimulcastEncoderAdapter. Bug: none Change-Id: I8182b2ab43f0c330ebdf077e9f7cbc79247da90e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202246 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33050}
2021-01-20 15:36:13 +01:00
"../rtc_base/experiments:encoder_info_settings",
Add settings to turn off VP8 base layer qp limit This quality boost means that we sometimes drop a _lot_ of frames in the base layer. It also interacts poorly with the bitrate adjuster since even if frames are dropped they are often over-sized. The setting still leaves the current behavior as default, but can be changed using the WebRTC-VideoRateControl field trial. Bug: webrtc:10155 Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367 Reviewed-on: https://webrtc-review.googlesource.com/c/122840 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26659}
2019-02-13 10:49:37 +01:00
"../rtc_base/experiments:rate_control_settings",
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 8). 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). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 11:33:59 +02:00
"../rtc_base/system:rtc_export",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
"../system_wrappers",
Refactors SimulcastEncoder Adapter. This done in preparation of VP9 support. Bug: webrtc:12354 Change-Id: Iabd220f9c7af2694374be1fc0f0de9a2deda3470 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201386 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32998}
2021-01-14 16:37:26 +01:00
"//third_party/abseil-cpp/absl/algorithm:container",
Propagate Environment through SimulcastEncoderAdapter when provided Bug: webrtc:15860 Change-Id: Iabd7752ada2f8f774de1e2adc02a4157004bf43c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342720 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41893}
2024-03-12 16:58:18 +01:00
"//third_party/abseil-cpp/absl/base:nullability",
Refactors SimulcastEncoder Adapter. This done in preparation of VP9 support. Bug: webrtc:12354 Change-Id: Iabd220f9c7af2694374be1fc0f0de9a2deda3470 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201386 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32998}
2021-01-14 16:37:26 +01:00
]
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -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_internal_video_codecs") {
Revert "Revert "Make some more targets publicly visible"" This reverts commit 55d1809d0d73592a1ddf4f0fb02ce7444fa066aa. Reason for revert: This cl was not the culprit for breaking chrome content/renderer deps. Original change's description: > Revert "Make some more targets publicly visible" > > This reverts commit 60d179256213c7516808aff827637cab8a47de89. > > 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: > > Make some more targets publicly visible > > > > To fix build errors introduced by > > https://webrtc-review.googlesource.com/c/src/+/24140 > > > > BUG=webrtc:8254 > > NOTRY=true > > > > Change-Id: I9cdf9cee39735368af78291134dbad70aebb7195 > > Reviewed-on: https://webrtc-review.googlesource.com/38660 > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21552} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I475ac382218fa77d33abc595f0773275d715a28e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38740 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21554} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: If8e9b7b1c934ec4b5ed61941c845e62e43bef97e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38841 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21559}
2018-01-10 15:57:32 +00:00
visibility = [ "*" ]
Break out the part of the iSAC codec that's used for Voice Activity Detection The audio processing code is using parts of the iSAC codec to do voice activity detection (VAD), but it's undesirable for it to pull in the entire iSAC codec as a dependency. So this CL factors out the parts of iSAC that's needed for VAD to a separate build target. Bug: webrtc:8396 Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e Reviewed-on: https://webrtc-review.googlesource.com/69640 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23110}
2018-04-27 04:31:53 +02:00
allow_poison = [ "software_video_codecs" ]
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
defines = []
libs = []
Add untracked headers to video_coding. This creates a new target for pure defines and interfaces. I think that makes sense (though include/ makes it harder to see when .cc and .h files should live together). Bug: webrtc:7620 Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e Reviewed-on: https://webrtc-review.googlesource.com/34657 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21516}
2018-01-08 11:05:10 +01:00
deps = [
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":codec",
":media_constants",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
":rtc_simulcast_encoder_adapter",
Propagate webrtc::Environment through MultiplexDecoderAdapter Bug: webrtc:15791 Change-Id: Ibe8fdc45722409b2cf6608ea6d8da2ea7e3472c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338621 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41747}
2024-02-09 12:43:15 +01:00
"../api/environment",
[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:encoded_image",
Directly include VideoBitrateAllocation in media/ targets Bug: webrtc:9271 Change-Id: I11a79c350a9de6edee203c9711ca97e266049f32 Reviewed-on: https://webrtc-review.googlesource.com/76943 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23368}
2018-05-22 15:37:23 +02:00
"../api/video:video_bitrate_allocation",
Move VideoCodecType from common_types.h to api/video/video_codec_type.h Bug: webrtc:7660 Change-Id: I9381364a64113dbb622b26acbf2b71228c3c4b96 Reviewed-on: https://webrtc-review.googlesource.com/c/111480 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25710}
2018-11-20 11:06:58 +01:00
"../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",
Make rtc_software_fallback_wrappers target visible. Need to depend on them from Chromium. Bug: webrtc:7925 Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17 Reviewed-on: https://webrtc-review.googlesource.com/82061 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23580}
2018-06-12 11:15:56 +02:00
"../api/video_codecs:rtc_software_fallback_wrappers",
Reland "Set default scalability mode for H.265 to L1T1." This is a reland of commit 775639e930f14a619974944594b40c633cc574a3 Original change's description: > Set default scalability mode for H.265 to L1T1. > > H.265 does not have software fallback, and it may have issue supporting > more than 1 temporal layers on some devices. Set default to L1T1 when > scalability is not configured, or if a scalability mode is reported as > not supported by encoder. > > Bug: chromium:41480904 > Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> > Cr-Commit-Position: refs/heads/main@{#43389} Bug: chromium:41480904 Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43412}
2024-11-14 20:51:07 +08:00
"../api/video_codecs:scalability_mode",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
"../api/video_codecs:video_codecs_api",
Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." This reverts commit f8c81ca4690aa3e470cc61633f512de383c308a8. Reason for revert: <Prepare to reland> Original change's description: > Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." > > This reverts commit 256733c78af655029cb04afae2c404afb41ea685. > > Reason for revert: <breaks downstream> > > Original change's description: > > Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate. > > > > Bug: webrtc:13573 > > Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36940} > > Bug: webrtc:13573 > Change-Id: I3341b6b96a56de63058c38943611b8c1629294ce > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262941 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36942} Bug: webrtc:13573 Change-Id: Iaf1222c58a18f378df20e4f83262b9a9da491712 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262943 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36985}
2022-05-24 10:32:32 +02:00
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
"../call:call_interfaces",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../modules/video_coding:video_codec_interface",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
"../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Export symbols needed by the Chromium component build (part 3). 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: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94 Reviewed-on: https://webrtc-review.googlesource.com/c/106343 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 14:13:50 +02:00
"../rtc_base/system:rtc_export",
Reland "Add dav1d decoder to WebRTC." This reverts commit 8498b7e7f6b90fa036de2a6887d34256f0565b4f. Reason for revert: Updating CL to include conditional build flag. Original change's description: > Revert "Add dav1d decoder to WebRTC." > > This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3. > > Reason for revert: High binary size increase > > Original change's description: > > Add dav1d decoder to WebRTC. > > > > Bug: none > > Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504 > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35394} > > TBR=danilchap@webrtc.org,mbonadei@webrtc.org,ilnik@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,ssilkin@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: none > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35395} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: Iff51848731646159e87e075c38af7cb6355f5b5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238661 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35409}
2021-11-23 11:00:24 +01:00
"../system_wrappers:field_trial",
Reland "Add a high bitrate full stack test with fake codec" In this reland, I disabled high bitrate webrtc perf test on Android32. This is a reland of 15df2774f4e85cf8900768c1793edcf17d651dcd Original change's description: > This CL adds a fake codec factory in WebRTC that can be used in tests to > produce target bitrate output. > We also add a high bitrate test that makes use of fake codec. This test assumes > ideal network conditions with target bandwidth being available and exercises > WebRTC calls with a high target bitrate(100 Mbps) end-to-end. TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org Bug: chromium:879723 Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9 Reviewed-on: https://webrtc-review.googlesource.com/c/117980 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26285}
2019-01-16 15:07:56 -05:00
"../test:fake_video_codecs",
Reland "Set default scalability mode for H.265 to L1T1." This is a reland of commit 775639e930f14a619974944594b40c633cc574a3 Original change's description: > Set default scalability mode for H.265 to L1T1. > > H.265 does not have software fallback, and it may have issue supporting > more than 1 temporal layers on some devices. Set default to L1T1 when > scalability is not configured, or if a scalability mode is reported as > not supported by encoder. > > Bug: chromium:41480904 > Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> > Cr-Commit-Position: refs/heads/main@{#43389} Bug: chromium:41480904 Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43412}
2024-11-14 20:51:07 +08:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
]
Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." This reverts commit f8c81ca4690aa3e470cc61633f512de383c308a8. Reason for revert: <Prepare to reland> Original change's description: > Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." > > This reverts commit 256733c78af655029cb04afae2c404afb41ea685. > > Reason for revert: <breaks downstream> > > Original change's description: > > Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate. > > > > Bug: webrtc:13573 > > Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36940} > > Bug: webrtc:13573 > Change-Id: I3341b6b96a56de63058c38943611b8c1629294ce > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262941 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36942} Bug: webrtc:13573 Change-Id: Iaf1222c58a18f378df20e4f83262b9a9da491712 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262943 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36985}
2022-05-24 10:32:32 +02:00
if (enable_libaom) {
defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
deps += [
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
]
}
Reland "Add dav1d decoder to WebRTC." This reverts commit 8498b7e7f6b90fa036de2a6887d34256f0565b4f. Reason for revert: Updating CL to include conditional build flag. Original change's description: > Revert "Add dav1d decoder to WebRTC." > > This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3. > > Reason for revert: High binary size increase > > Original change's description: > > Add dav1d decoder to WebRTC. > > > > Bug: none > > Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504 > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35394} > > TBR=danilchap@webrtc.org,mbonadei@webrtc.org,ilnik@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,ssilkin@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: none > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35395} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: Iff51848731646159e87e075c38af7cb6355f5b5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238661 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35409}
2021-11-23 11:00:24 +01:00
if (rtc_include_dav1d_in_internal_decoder_factory) {
deps += [ "../modules/video_coding/codecs/av1:dav1d_decoder" ]
}
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
sources = [
"engine/fake_video_codec_factory.cc",
"engine/fake_video_codec_factory.h",
"engine/internal_decoder_factory.cc",
"engine/internal_decoder_factory.h",
"engine/internal_encoder_factory.cc",
"engine/internal_encoder_factory.h",
]
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
}
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_audio_video") {
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.
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
defines = []
libs = []
deps = [
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":audio_source",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":codec",
":media_channel",
Remove AsSendChannel/AsReceiveChannel methods This CL removes a couple more opportunities for client code to interact directly with the MediaChannel implementation classes. No-try because of infra failure. No-Try: true Bug: webrtc:13931 Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38935}
2022-12-16 14:46:31 +00:00
":media_channel_impl",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":media_engine",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
":rid_description",
":rtc_media_config",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":rtp_utils",
":stream_params",
Update IWYU mappings with a few more lines. Also run IWYU on a file picked for testing and check in the result. Bug: none Change-Id: Ide36bc59d126064f2bab7af441f72a6e8477c848 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280601 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38481}
2022-10-26 13:51:35 +00:00
"../api:array_view",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:audio_options_api",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:call_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../api:field_trials_view",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:frame_transformer_interface",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:libjingle_peerconnection_api",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:make_ref_counted",
Implement new specification for degradation preference The degradation preference is now based on the content hint of the track if it's unspecified. Bug: webrtc:11164 Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30691}
2020-03-05 13:39:55 +01:00
"../api:media_stream_interface",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api:priority",
"../api:rtc_error",
"../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",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api: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",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:transport_api",
Move webrtc::AudioDeviceModule include to api/ folder Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-19 19:46:03 +00:00
"../api/audio:audio_device",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/audio:audio_frame_api",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../api/audio:audio_frame_processor",
In media engine replace forward declaration with proper includes Add missing includes and dependency. Forward declaring VideoDecoderFactory started causing compilation errors when user doesn't explicitly specify one. Bug: None Change-Id: Id7d1634d5dc409a1c3f9e4778deee21f345e7cdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139250 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28124}
2019-05-31 15:51:04 +02:00
"../api/audio:audio_mixer_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
"../api/audio:audio_processing_statistics",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/audio_codecs:audio_codecs_api",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
Reland "in WebrtcVoiceEngine allow to set TaskQueueFactory" in production code keep using GlobalTaskQueueFactory() in tests switch to use DefaultTaskQueueFactory directly. This reverts commit e27ccf9a1681e0e4ff9281f9a18fea357d2bc890. Reason for revert: addressed the failure with patchset#2 Original change's description: > Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory" > > This reverts commit a39254da593bbdb0b1e072a44827229680afe3ee. > > Reason for revert: Tests are failing due to ThreadChecker's called on valid thread. > > Original change's description: > > in WebrtcVoiceEngine allow to set TaskQueueFactory > > > > in production code keep using GlobalTaskQueueFactory() > > in tests switch to use DefaultTaskQueueFactory directly. > > > > Bug: webrtc:10284 > > Change-Id: I170274a98324796623089a965a39f0cbb7e281d9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128878 > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27296} > > TBR=danilchap@webrtc.org,steveanton@webrtc.org > > Change-Id: I9742e5d0171a94f3840e197c40fdb44523e4963b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10284 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129780 > Reviewed-by: Amit Hilbuch <amithi@webrtc.org> > Commit-Queue: Amit Hilbuch <amithi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27297} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10284 Change-Id: I55fd5811c68d04c3e8cf537974496460b38c1d4f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129933 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27348}
2019-03-27 18:51:45 +01:00
"../api/task_queue",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Remove api/bitrate_constraints.h. Bug: webrtc:8733 Change-Id: Iaeb26e07d399f25dc18b0c4af38ed400577a5d3a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153220 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29217}
2019-09-17 14:47:38 +02:00
"../api/transport:bitrate_settings",
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",
Extend WebRTC-Video-MinVideoBitrate to experiment per-codec The experiment was extended to support per-codec minimum bitrates for the following codecs: * VP8 * VP9 * H.264 The old semantic meaning for the field trial is retained, in that specifying "br:" applies a minimum bitrate to all codecs. If "br:" is not specified, the per-codec minimum config is consulted. Bug: webrtc:11024 Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29450}
2019-10-11 16:19:43 +02:00
"../api/units:data_rate",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:recordable_encoded_frame",
"../api/video:resolution",
[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:video_bitrate_allocation",
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 10:02:56 -08:00
"../api/video:video_bitrate_allocator_factory",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video:video_codec_constants",
"../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",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/video:video_stream_encoder",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video_codecs:rtc_software_fallback_wrappers",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../api/video_codecs:scalability_mode",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../call:payload_type",
"../call:payload_type_picker",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../call:receive_stream_interface",
"../call:rtp_interfaces",
Split up the call/video_stream_api target The split shows that some places don't need it at all. Most other places will depend on both send and receive stream targets. Bug: webrtc:373151158 Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43230}
2024-10-13 21:08:30 +00:00
"../call:video_receive_stream_api",
"../call:video_send_stream_api",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../common_video",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../common_video:frame_counts",
Async audio processing API API to injecting a heavy audio processing operation into WebRTC audio capture pipeline Bug: webrtc:12003 Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32291}
2020-09-30 18:27:39 +02:00
"../modules/async_audio_processing:async_audio_processing",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../modules/audio_device",
"../modules/audio_device:audio_device_impl",
"../modules/audio_mixer:audio_mixer_impl",
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/aec_dump",
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
2019-10-15 10:10:26 +02:00
"../modules/audio_processing/agc:gain_control_interface",
stats: implement flexfec fecBytesReceived stats for FlexFEC specified in https://github.com/w3c/webrtc-stats/pull/762 and take FlexFEC into account for receive statistics. BUG=webrtc:15250 Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40325}
2023-06-21 13:53:41 +02:00
"../modules/rtp_rtcp",
Move helpers to parse base rtp packet fields to rtp_rtcp module rtp_rtcp_format is lighter build target than rtc_media_base and a more natural place to keep rtp parsing functions. Bug: None Change-Id: Ibcb5661cc65edbdc89a63f3e411d7ad1218353cc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226330 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34504}
2021-07-19 13:20:46 +00:00
"../modules/rtp_rtcp:rtp_rtcp_format",
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/video_coding",
Complete moving video_coding headers. Bug: webrtc:7620 Change-Id: Ic553cd083ef267b19897777340120b1f3e4765f2 Reviewed-on: https://webrtc-review.googlesource.com/39663 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21633}
2018-01-16 10:11:40 +01:00
"../modules/video_coding:video_codec_interface",
Generalize SimulcastEncoderAdapter, use for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 16:16:38 +02:00
"../modules/video_coding:video_coding_utility",
Reland "When VP9 SVC is used, use SvcConfig to set max bitrate for the stream." This is a reland of commit 3afb8e24311dc1297150d4011894b6cb00841735 Patchset 1 is the original CL. Patchset 2 contains a fix: Depending on call site, the number of spatial layers for VP9 might be signalled in three different ways. One of them was afaict only used in out perf tests, and resulted in the max bitrate being incorrectly capped. The fix now checks that field too. Original change's description: > When VP9 SVC is used, use SvcConfig to set max bitrate for the stream. > > Currently, a default max bitrate is determined within WebRtcVideoEngine, > which maxes out at 2.5Mbps - and that limits the max bitrate deteremined > by SvcConfig for resolutions above 720p. > > This does not affect simulcast, as WebRtcVideoEngine already knows to > trust the rate allocation in simulcast.cc instead. > > Bug: webrtc:14017 > Change-Id: I0c310a6fd496e9e5a10eae45838900068aa1ae2d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267160 > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#37370} Bug: webrtc:14017, webrtc:14234 Change-Id: Idcaf4321a20c917e4049522c577336ddcfc7ffbb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267860 Auto-Submit: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37446}
2022-07-05 13:02:28 +02:00
"../modules/video_coding:webrtc_vp9_helpers",
New enum ScalabilityMode. Used instead of string representation in lower-levels of encoder configuration, to avoid string comparisons (with risk of misspelling) in lots of places. Bug: webrtc:11607 Change-Id: I4d51c2265aac297c29976d2aa601d8ffb33b7326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259870 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36706}
2022-04-29 11:03:13 +02:00
"../modules/video_coding/svc:scalability_mode_util",
This CL removes all usages of our custom ostream << overloads. This prepares us for removing them altogether. Bug: webrtc:8982 Change-Id: I66002cc8d4bf0e07925766d568d2498422f0f38e Reviewed-on: https://webrtc-review.googlesource.com/64142 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Henrik Grunell <henrikg@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22707}
2018-04-03 13:40:05 +02:00
"../rtc_base:audio_format_to_string",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../rtc_base:dscp",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Move race_checker out of rtc_base_approved Bug: webrtc:9838 Change-Id: If180abcca1ef598314de3aed70e4a6eb04f062d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258770 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36607}
2022-04-14 13:38:05 +02:00
"../rtc_base:race_checker",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../rtc_base:safe_conversions",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../rtc_base:socket",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:ssl",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../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",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../rtc_base/experiments:field_trial_parser",
Extend WebRTC-Video-MinVideoBitrate to experiment per-codec The experiment was extended to support per-codec minimum bitrates for the following codecs: * VP8 * VP9 * H.264 The old semantic meaning for the field trial is retained, in that specifying "br:" applies a minimum bitrate to all codecs. If "br:" is not specified, the per-codec minimum config is consulted. Bug: webrtc:11024 Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29450}
2019-10-11 16:19:43 +02:00
"../rtc_base/experiments:min_video_bitrate_experiment",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../rtc_base/experiments:normalize_simulcast_size_experiment",
Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. - Centralize field trial string reading to RateControlSettings - Cache RateControlSettings at all production code use sites Bug: None Change-Id: I0dbce9cc97fea0bc780982e7ef270b417a8c15bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158664 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29680}
2019-10-30 13:01:46 +01:00
"../rtc_base/experiments:rate_control_settings",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../rtc_base/network:sent_packet",
Migrate media/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I69e4a1b37737ac8dd852a032612623c4c4f3a30b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176744 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31648}
2020-07-07 15:43:11 +02:00
"../rtc_base/synchronization:mutex",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../rtc_base/system:file_wrapper",
Updated associated payload types without recreating receive streams. Bug: webrtc:11993 Change-Id: I49c61653b296b1b3ca6a12fa75ac699ee58f096c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271543 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37799}
2022-08-16 10:23:47 +02:00
"../rtc_base/system:no_unique_address",
Export symbols needed by the Chromium component build (part 3). 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: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94 Reviewed-on: https://webrtc-review.googlesource.com/c/106343 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 14:13:50 +02:00
"../rtc_base/system:rtc_export",
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",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../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:metrics",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"../video/config:encoder_config",
"//third_party/abseil-cpp/absl/algorithm",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Change SSRC-passing for MediaChannel from external to callback This makes the handling somewhat more uniform, and is the same for both video and audio channels. Bug: webrtc:13931 Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40107}
2023-05-22 12:49:05 +00:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/functional:any_invocable",
Use parsed packet from RtpTransport::DemuxPacket in engine and call With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused. Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid. Bug: webrtc:7135, webrtc:14795 Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39058}
2023-01-10 14:28:25 +01:00
"//third_party/abseil-cpp/absl/functional:bind_front",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"//third_party/abseil-cpp/absl/strings",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
]
sources = [
"engine/adm_helpers.cc",
"engine/adm_helpers.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
"engine/webrtc_media_engine.cc",
"engine/webrtc_media_engine.h",
"engine/webrtc_video_engine.cc",
"engine/webrtc_video_engine.h",
"engine/webrtc_voice_engine.cc",
"engine/webrtc_voice_engine.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
]
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 = []
Delete interface class VideoCaptureExternal Also delete corresponding and unused create method VideoCaptureFactory::Create(VideoCaptureExternal...), the code under modules/video_capture/external, and the build target modules/video_capture:video_capture. Bug: None Change-Id: I5ec6139e9ecf460f93ede847868f7f80dbc019f5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131385 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27506}
2019-04-09 08:24:58 +02:00
if (!build_with_chromium) {
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
public_configs += [ ":rtc_media_defines_config" ]
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
}
Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ ) Reason for revert: Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'". *Now* the internal projects are fixed and the fix is verified. Original issue's description: > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ ) > > Reason for revert: > Reverting again: internal project issues were apparently not completely fixed. > > Original issue's description: > > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ ) > > > > Reason for revert: > > Revert the revert now that internal projects are updated. > > > > Original issue's description: > > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ ) > > > > > > Reason for revert: > > > Breaks internal project. > > > > > > Original issue's description: > > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'. > > > > > > > > A low priority task queue is added to WebRTCVoiceEngine. The > > > > start/stop debug calls make file logging happen on the task queue. > > > > > > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory, > > > > so that it can be shared for low priority tasks between different > > > > subcomponents. It will require some changes to MediaEngine, > > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal > > > > projects. > > > > > > > > A task queue must be created and destroyed from the same thread. With > > > > this CL that will be the worker thread, which creates and destroys > > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to > > > > the signaling thread. > > > > > > > > NOTRY=True # tests just passed > > > > > > > > BUG=webrtc:7404 > > > > > > > > Review-Url: https://codereview.webrtc.org/2896813002 > > > > Cr-Commit-Position: refs/heads/master@{#18252} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/c61bf947b4ac31f3500858ffcae6fee39d799930 > > > > > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7404 > > > > > > Review-Url: https://codereview.webrtc.org/2904893002 > > > Cr-Commit-Position: refs/heads/master@{#18255} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/be68b72cfad0686dcd892bba1368b199a7ee16ca > > > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:7404 > > > > Review-Url: https://codereview.webrtc.org/2903153005 > > Cr-Commit-Position: refs/heads/master@{#18270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/d2303a2338106feab684860f1c133877b46bdd4f > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7404 > > Review-Url: https://codereview.webrtc.org/2910633002 > Cr-Commit-Position: refs/heads/master@{#18272} > Committed: https://chromium.googlesource.com/external/webrtc/+/fe9ecb07ea8254d8a09605f25203a4d045b3ffee TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2904423002 Cr-Commit-Position: refs/heads/master@{#18300}
2017-05-29 02:56:27 -07:00
if (rtc_enable_protobuf) {
Add RtpEncodingParameters.adaptive_ptime. When enabled: - Creates an audio network adapter config that is passed to audio send stream. - Configures a lower default min bitrate. All parameters can be configured via a field trial that can also force enable the audio network adaptor (this is mainly intended for testing). Bug: chromium:1086942 Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31565}
2020-06-25 14:09:58 +02:00
deps += [
"../modules/audio_coding:ana_config_proto",
"../modules/audio_processing/aec_dump:aec_dump_impl",
]
Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ ) Reason for revert: Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'". *Now* the internal projects are fixed and the fix is verified. Original issue's description: > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ ) > > Reason for revert: > Reverting again: internal project issues were apparently not completely fixed. > > Original issue's description: > > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ ) > > > > Reason for revert: > > Revert the revert now that internal projects are updated. > > > > Original issue's description: > > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ ) > > > > > > Reason for revert: > > > Breaks internal project. > > > > > > Original issue's description: > > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'. > > > > > > > > A low priority task queue is added to WebRTCVoiceEngine. The > > > > start/stop debug calls make file logging happen on the task queue. > > > > > > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory, > > > > so that it can be shared for low priority tasks between different > > > > subcomponents. It will require some changes to MediaEngine, > > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal > > > > projects. > > > > > > > > A task queue must be created and destroyed from the same thread. With > > > > this CL that will be the worker thread, which creates and destroys > > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to > > > > the signaling thread. > > > > > > > > NOTRY=True # tests just passed > > > > > > > > BUG=webrtc:7404 > > > > > > > > Review-Url: https://codereview.webrtc.org/2896813002 > > > > Cr-Commit-Position: refs/heads/master@{#18252} > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/c61bf947b4ac31f3500858ffcae6fee39d799930 > > > > > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > > > # Skipping CQ checks because original CL landed less than 1 days ago. > > > NOPRESUBMIT=true > > > NOTREECHECKS=true > > > NOTRY=true > > > BUG=webrtc:7404 > > > > > > Review-Url: https://codereview.webrtc.org/2904893002 > > > Cr-Commit-Position: refs/heads/master@{#18255} > > > Committed: https://chromium.googlesource.com/external/webrtc/+/be68b72cfad0686dcd892bba1368b199a7ee16ca > > > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > > # Not skipping CQ checks because original CL landed more than 1 days ago. > > BUG=webrtc:7404 > > > > Review-Url: https://codereview.webrtc.org/2903153005 > > Cr-Commit-Position: refs/heads/master@{#18270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/d2303a2338106feab684860f1c133877b46bdd4f > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7404 > > Review-Url: https://codereview.webrtc.org/2910633002 > Cr-Commit-Position: refs/heads/master@{#18272} > Committed: https://chromium.googlesource.com/external/webrtc/+/fe9ecb07ea8254d8a09605f25203a4d045b3ffee TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7404 Review-Url: https://codereview.webrtc.org/2904423002 Cr-Commit-Position: refs/heads/master@{#18300}
2017-05-29 02:56:27 -07:00
} else {
deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
}
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
}
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
rtc_source_set("rtc_data_sctp_transport_internal") {
sources = [ "sctp/sctp_transport_internal.h" ]
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
deps = [
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":media_channel",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Update SctpTransportInternal to use RTCError. This avoids a couple of layers of error code conversion, reduces dependency on cricket error types and allows us to preserve error information from dcsctp. Along the way remove SendDataResult. Bug: none Change-Id: I1ad18a8f0b2fb181745b19c49f36f270708720c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298305 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39619}
2023-03-21 14:48:51 +01:00
"../api:rtc_error",
datachannel: Merge SendDataParams and DMT types with webrtc equivalent cricket::SendDataParams is replaced by webrtc::SendDataParams. cricket::DataMessageType is replaced by webrtc::DataMessageType. The sid member from cricket::SendDataParams is now passed as an argument to functions that used one when necessary. Bug: webrtc:7484 Change-Id: Ia4a89c9651fb54ab9a084a6098d49130b6319e1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217761 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33966}
2021-05-10 11:29:56 +02:00
"../api/transport:datagram_transport_interface",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Reland "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",
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
]
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
}
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
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
if (rtc_build_dcsctp) {
rtc_library("rtc_data_dcsctp_transport") {
sources = [
"sctp/dcsctp_transport.cc",
"sctp/dcsctp_transport.h",
]
deps = [
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":media_channel",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
":rtc_data_sctp_transport_internal",
"../api:array_view",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../api:dtls_transport_interface",
pc: Remove additional buffering in SctpDataChannel This CL removes the send buffers (but not the receive buffer) from SctpDataChannel and increases the send buffer in DcSctpSocket instead. The reasons are: 1) Simplify the code. This additional buffering was strictly needed before we migrated away from usrsctp, as that send buffer was very limited in size (by design). But with the migration to dcSCTP, it's no longer needed, so it just adds complexity. 2) Make `RTCDataChannel::bufferedAmount` correct. Before this CL, it represented just the data buffered in SctpDataChannel, and not the data accepted by the SCTP socket, but not yet put on the wire. This makes it hard for clients to know when a message has ever been sent. 3) Better handle draining data on data channel close. While this is not implemented in dcSCTP, having a single buffer makes this easier to add. While most of this CL is straightforward, the handling of bufferedAmount in the signaling thread (in RTCDataChannel in Blink), is a bit special. The number returned by `RTCDataChannel::bufferedAmount` is not what the true value is inside the SCTP socket, but an eventual consistent view of that value. When a message is sent, the value is incremented and: - Before this change: When a message was put on the SCTP socket, the view's value was decremented. Which made the view reflect what was buffered outside the SCTP socket, and that buffering is now gone. - After this change: SctpDataChannel will track what RTCDataChannel will think it is, and provide updates to that number as we are notified that it's reduced - by setting a "low threshold" callback trigger. A bonus with the new behavior is that it will be eventually consistent and auto-heal also in error conditions - when messages are dropped due to errors (bad input, bad state, etc). Previously, the bufferedAmount value could drift away from the correct value on errors. Note that a big chunk of unit tests were removed with this CL, as those tested how the buffering behaved. Now, there is no buffering, so the removed test cases represent a simpler interface. This CL has been extensively tested with data channel benchmarks that use the bufferedAmount thresholds (in Javascript). Bug: chromium:40072842 Change-Id: I1a6a4af6b6e1116832f5028f989ce9f44683d229 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343361 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41945}
2024-03-18 13:51:40 +01:00
"../api:libjingle_peerconnection_api",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../api:rtc_error",
"../api:sequence_checker",
Keep Environment instead of test field trials in FakeCall test object To pass field trials to EncoderStreamFactory in FakeVideoSendStream and thus reduce dependency on the global field trial. Bug: webrtc:10335 Change-Id: Iad32881c2d9158fe1d77f1b71f8d606374ea111e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42023}
2024-04-08 18:35:40 +02:00
"../api/environment",
Move to_queued_task.h and pending_task_safety_flag.h into public API Bug: b/235812579 Change-Id: I9fa3dc4a65044df8b44fff4e9bfeac7233fa381c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266080 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37248}
2022-06-16 21:27:45 +02:00
"../api/task_queue:pending_task_safety_flag",
Allow specifying delayed task precision of dcsctp::Timer. Context: The timer precision of PostDelayedTask() is about to be lowered to include up to 17 ms leeway. In order not to break use cases that require high precision timers, PostDelayedHighPrecisionTask() will continue to have the same precision that PostDelayedTask() has today. webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which precision to use when calling PostDelayedTaskWithPrecision(). See go/postdelayedtask-precision-in-webrtc for motivation and a table of delayed task use cases in WebRTC that are "high" or "low" precision. Most timers in DCSCTP are believed to only be needing low precision (see table), but the delayed_ack_timer_ of DataTracker[1] is an example of a use case that is likely to break if the timer precision is lowered (if ACK is sent too late, retransmissions may occur). So this is considered a high precision use case. This CL makes it possible to specify the precision of dcsctp::Timer. In a follow-up CL we will update delayed_ack_timer_ to kHigh precision. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340 Bug: webrtc:13604 Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081 Reviewed-by: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35809}
2022-01-26 18:38:13 +01:00
"../api/task_queue:task_queue",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../api/transport:datagram_transport_interface",
dcsctp: Add DcSctpSocketFactory The factory allows us to isolate the implementation from users who only need to depend directly on the public folder now. Bug: webrtc:12614 Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34330}
2021-06-01 18:39:49 +02:00
"../net/dcsctp/public:factory",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
"../net/dcsctp/public:socket",
"../net/dcsctp/public:types",
dcsctp: Extract logging packet observer as utility It is useful for more than just the transport. Bug: webrtc:12961 Change-Id: Iad064c8fb707ca589a1c232e17436338fb06623d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225543 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34451}
2021-06-19 07:59:01 +02:00
"../net/dcsctp/public:utils",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
"../net/dcsctp/timer:task_queue_timeout",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../p2p:dtls_transport_internal",
Refactor a lot of the p2p:rtc_p2p target This CL splits many of the source files in p2p:rtc_p2p into individual compile targets. One target - connection_and_port - was left with multiple source files because it was too tangled to detangle at once. Bug: webrtc:15796 Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41611}
2024-01-25 12:27:18 +00:00
"../p2p:packet_transport_internal",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Move event_tracer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-19 17:47:13 +02:00
"../rtc_base:event_tracer",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove sigslot usage from SctpTransportInternal Bug: webrtc:11943 Change-Id: I42edf8e2e15e580bcda090447a7aae4a56366b33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270661 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37867}
2022-08-22 11:34:29 +02:00
"../rtc_base:macromagic",
Move random out of rtc_base_approved Bug: webrtc:9838 Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-14 12:41:26 +02:00
"../rtc_base:random",
dcsctp: Add API to indicate packet send status Before this change, there was no way for a client to indicate to the dcSCTP library if a packet that was supposed to be sent, was actually sent. It was assumed that it always was. To handle temporary failures better, such as retrying to send packets that failed to be sent when the send buffer was full, this information is propagated to the library. Note that this change only covers the API and adaptations to clients. The actual implementation to make use of this information is done as a follow-up change. Bug: webrtc:12943 Change-Id: I8f9c62e17f1de1566fa6b0f13a57a3db9f4e7684 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228563 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34767}
2021-08-12 15:21:25 +02:00
"../rtc_base:socket",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../rtc_base:stringutils",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
"../rtc_base:threading",
sctp: Handle concurrent data channel reset in transport The state machine for handling resets couldn't handle resets happening from both sides at the same time. Bug: webrtc:13994 Change-Id: I2c268e54f4c5c9858913faef91ff00f6af956e99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261305 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36799}
2022-05-05 23:43:44 +02:00
"../rtc_base/containers:flat_map",
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback Instead of using PacketTransportInternal::SignalReadPacket. Bug: webrtc:15368 Change-Id: Icdc2d7f85df6db944f0ba0232891e6c5a8986a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340440 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41823}
2024-02-27 09:01:15 +01:00
"../rtc_base/network:received_packet",
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
"../rtc_base/third_party/sigslot:sigslot",
"../system_wrappers",
"//third_party/abseil-cpp/absl/strings:strings",
]
}
}
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
rtc_library("rtc_data_sctp_transport_factory") {
defines = []
sources = [
"sctp/sctp_transport_factory.cc",
"sctp/sctp_transport_factory.h",
]
deps = [
":rtc_data_sctp_transport_internal",
sctp: Pass webrtc::Environment to DcSctpTransport The DcSctpTransport will soon use field trials to conditionally enable some options. And overall, there is a migration project to start using the Environment and this CL is in that direction, also setting the boundary; The dcSCTP library should not depend on it. But the transport is allowed to. Bug: webrtc:14997 Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41872}
2024-02-28 12:16:15 +01:00
"../api/environment",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../api/transport:sctp_transport_factory_interface",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../p2p:dtls_transport_internal",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../rtc_base:threading",
"../rtc_base/system:unused",
]
if (rtc_enable_sctp) {
Remove usrsctp, dcSCTP is now the unique SCTP implementation Bug: chromium:1243702 Change-Id: Id11299d26f0f8713a57781b57277837aace531f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251821 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36423}
2022-03-31 19:15:10 +02:00
assert(rtc_build_dcsctp, "An SCTP backend is required to enable SCTP")
sctp: Add DcsctpTransport based on dcSCTP Bug: webrtc:12614 Change-Id: Ie710621610fff9f8bb6c7d800419675892d6a70c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33935}
2021-05-06 10:50:07 +02:00
}
if (rtc_build_dcsctp) {
defines += [ "WEBRTC_HAVE_DCSCTP" ]
deps += [
":rtc_data_dcsctp_transport",
"../system_wrappers",
"../system_wrappers:field_trial",
]
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
}
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_media") {
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.
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
deps = [ ":rtc_audio_video" ]
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
}
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
if (rtc_include_tests) {
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_media_tests_utils") {
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
testonly = true
Remove dependency in FakeWebRtcVideoCodecFactories. Previously, constructing a PeerConnection or WebRtcVideoEngine with fake encoder/decoder factories would result in the real, built-in factories also being used. In https://webrtc-review.googlesource.com/c/src/+/71162, this changed, so to temporarily allow tests to continue working exactly the same as before, the fake factories started encapsulating the real factories. This CL removes that behavior and updates the tests accordingly. Bug: webrtc:9228 Change-Id: Ida14a1e3f5f5a0e2f03100b7895b3b1bdf0a0a42 Reviewed-on: https://webrtc-review.googlesource.com/75260 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23209}
2018-05-14 09:48:06 +02:00
defines = []
Enable GN check for webrtc/media BUG=webrtc:6828, webrtc:7245 NOTRY=True Review-Url: https://codereview.webrtc.org/2716143002 Cr-Commit-Position: refs/heads/master@{#17067}
2017-03-06 06:04:55 -08:00
deps = [
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":audio_source",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":codec",
":media_channel",
Split fake media channel classes This allows to remove some calls to CreateMediaChannel in the RtpTransceiver code. This removes the fake engines owning the channels and moves the responsibility to the tests themselves as it's quite hard to both return a unique_ptr to a channel and still own it. The various channel getters from the fake engine are thus also removed and tests updated accordingly, the channel is retrieved from internal structs in the tests by going through the RtpTransceiver objects as it's not possible to safely get the channels from only a sender or receiver. As some tests are running in both PlanB and Unified Plan, getting a transceiver is not working for PlanB. As PlanB has been deprecated and will eventually be removed, the problematic tests have either been removed or updated to only run with Unified Plan. Bug: webrtc:13931 Change-Id: I0571beca8b9ef2f2089d500802b7b124268d9de3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40366}
2023-06-26 00:26:07 +00:00
":media_channel_impl",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":media_engine",
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
":rtc_audio_video",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
":rtc_internal_video_codecs",
":rtc_media",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
":rtc_simulcast_encoder_adapter",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":rtp_utils",
":stream_params",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":video_common",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:call_api",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../api:fec_controller_api",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:frame_transformer_interface",
"../api:rtc_error",
"../api:rtp_headers",
Split fake media channel classes This allows to remove some calls to CreateMediaChannel in the RtpTransceiver code. This removes the fake engines owning the channels and moves the responsibility to the tests themselves as it's quite hard to both return a unique_ptr to a channel and still own it. The various channel getters from the fake engine are thus also removed and tests updated accordingly, the channel is retrieved from internal structs in the tests by going through the RtpTransceiver objects as it's not possible to safely get the channels from only a sender or receiver. As some tests are running in both PlanB and Unified Plan, getting a transceiver is not working for PlanB. As PlanB has been deprecated and will eventually be removed, the problematic tests have either been removed or updated to only run with Unified Plan. Bug: webrtc:13931 Change-Id: I0571beca8b9ef2f2089d500802b7b124268d9de3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40366}
2023-06-26 00:26:07 +00:00
"../api:rtp_parameters",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:rtp_sender_interface",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/adaptation:resource_adaptation_api",
"../api/audio:audio_frame_api",
Move webrtc::AudioProcessing include to api/ folder Bug: webrtc:15874 Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-19 15:07:08 +00:00
"../api/audio:audio_processing",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:frame_decryptor_interface",
Propagate Environment through SimulcastEncoderAdapter when provided Bug: webrtc:15860 Change-Id: Iabd7752ada2f8f774de1e2adc02a4157004bf43c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342720 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41893}
2024-03-12 16:58:18 +01:00
"../api/environment",
In FakeNetworkInterface drop dependency on rtc::MessageHandler Bug: webrtc:9702 Change-Id: Ida110c0a790ee737bcffff0dee95dd35787205e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271299 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37819}
2022-08-17 10:24:02 +02:00
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/transport:bitrate_settings",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
Implement FakeCall::DeliverRtpPacket and DeliverRtcpPacket This is done in peparation for using these methods in the engines. Bug: webrtc:7135, webrtc:14795 Change-Id: I1255c035437d31398327318c3dbd73e70a11a5cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290577 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39044}
2023-01-05 15:20:33 +01:00
"../api/units:timestamp",
[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:encoded_image",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video:video_bitrate_allocation",
"../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",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../call:mock_rtp_interfaces",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../call:payload_type",
"../call:payload_type_picker",
"../call:rtp_interfaces",
Split up the call/video_stream_api target The split shows that some places don't need it at all. Most other places will depend on both send and receive stream targets. Bug: webrtc:373151158 Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43230}
2024-10-13 21:08:30 +00:00
"../call:video_receive_stream_api",
"../call:video_send_stream_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
"../common_video",
"../modules/audio_processing",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../modules/rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 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@{#21108}
2017-12-06 07:51:33 +01:00
"../modules/rtp_rtcp:rtp_rtcp_format",
Add untracked headers to video_coding. This creates a new target for pure defines and interfaces. I think that makes sense (though include/ makes it harder to see when .cc and .h files should live together). Bug: webrtc:7620 Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e Reviewed-on: https://webrtc-review.googlesource.com/34657 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21516}
2018-01-08 11:05:10 +01:00
"../modules/video_coding:video_codec_interface",
Fix Gn Untracked headers in webrtc/modules/video_coding. Fixed following headers in this CL =================================== src/webrtc/modules/video_coding/sequence_number_util.h src/webrtc/modules/video_coding/codecs/interface/common_constants.h src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h src/webrtc/modules/video_coding/test/test_util.h src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h Remaining: =========== src/webrtc/modules/video_coding/include/video_codec_interface.h src/webrtc/modules/video_coding/include/video_error_codes.h BUG=webrtc:7620 Review-Url: https://codereview.webrtc.org/3012323002 Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 03:08:08 -07:00
"../modules/video_coding:video_coding_utility",
Add SVC fallback. Fallback to a default value if the scalability mode is unset or not supported by the codec. The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set). Bug: webrtc:11607 Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39074}
2023-01-11 15:50:58 +01:00
"../modules/video_coding/svc:scalability_mode_util",
Move buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-19 17:07:23 +02:00
"../rtc_base:buffer",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Move copy_on_write_buffer out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-19 17:40:46 +02:00
"../rtc_base:copy_on_write_buffer",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:dscp",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:gunit_helpers",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:network_route",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Fix circular dependencies in webrtc_common. One reason for the circular deps is that common_types.h is a historical dumping ground for various structs and defines that are believed to be generally useful. I tried moving things out that did not appear to be used downstream (StreamCounters, RtpCounters etc) and moved the things that seemed used (RtpHeader + supporting structs) to a new file api/rtp_headers.h. This makes their place in the api more clear while moving out the things that don't belong in the API in the first place. I had to extract out typedefs.h from webrtc_common to resolve another circular dependency. I believe checks includes typedefs, but common depends on checks. Bug: webrtc:7745 Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b Reviewed-on: https://webrtc-review.googlesource.com/33001 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:40:10 +01:00
"../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",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../rtc_base/network:sent_packet",
Migrate media/ to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I69e4a1b37737ac8dd852a032612623c4c4f3a30b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176744 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31648}
2020-07-07 15:43:11 +02:00
"../rtc_base/synchronization:mutex",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../rtc_base/third_party/sigslot",
Propagate environment to RtpSenders Will be later used to conditionally enable mixed codec simulcast with a field trial. Bug: webrtc:42220378 Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42929}
2024-09-03 09:13:23 +00:00
"../test:explicit_key_value_config",
"../test:scoped_key_value_config",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../test:test_support",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../video/config:encoder_config",
Create EncoderStreamFactory in VideoStreamEncoder This cl/ changes so that the EncoderStreamFactory is not created inside WebRtcVideoSendStream (webrtc_video_engine). The benifit of this is that the VideoStreamEncoder can then amend the EncoderStreamFactory with state (and types) w/o exposing it in VideoEncoderConfig. I.e as an alternative to changes done inside https://webrtc-review.googlesource.com/c/src/+/276742. The fake_webrtc_call is modified to (if needed) create it's own EncoderStreamFactory if needed. Note: this cl/ will have to be merged with with https://webrtc-review.googlesource.com/c/src/+/277002. Bug: webrtc:14451 Change-Id: I3d896b227d39725ba6409622e8d09d14bd45d5fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277160 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38237}
2022-09-28 16:28:42 +02:00
"../video/config:streams_config",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"//testing/gtest",
Use Abseil container algorithms in media/ Bug: None Change-Id: I292e3401bbf19a66271dd5ef2b3ca4f8dcfd155d Reviewed-on: https://webrtc-review.googlesource.com/c/120003 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26434}
2019-01-28 17:27:58 -08:00
"//third_party/abseil-cpp/absl/algorithm:container",
Split fake media channel classes This allows to remove some calls to CreateMediaChannel in the RtpTransceiver code. This removes the fake engines owning the channels and moves the responsibility to the tests themselves as it's quite hard to both return a unique_ptr to a channel and still own it. The various channel getters from the fake engine are thus also removed and tests updated accordingly, the channel is retrieved from internal structs in the tests by going through the RtpTransceiver objects as it's not possible to safely get the channels from only a sender or receiver. As some tests are running in both PlanB and Unified Plan, getting a transceiver is not working for PlanB. As PlanB has been deprecated and will eventually be removed, the problematic tests have either been removed or updated to only run with Unified Plan. Bug: webrtc:13931 Change-Id: I0571beca8b9ef2f2089d500802b7b124268d9de3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310340 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40366}
2023-06-26 00:26:07 +00:00
"//third_party/abseil-cpp/absl/functional:any_invocable",
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",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/strings:string_view",
Enable GN check for webrtc/media BUG=webrtc:6828, webrtc:7245 NOTRY=True Review-Url: https://codereview.webrtc.org/2716143002 Cr-Commit-Position: refs/heads/master@{#17067}
2017-03-06 06:04:55 -08:00
]
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
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
"base/fake_frame_source.cc",
"base/fake_frame_source.h",
"base/fake_media_engine.cc",
"base/fake_media_engine.h",
"base/fake_network_interface.h",
"base/fake_rtp.cc",
"base/fake_rtp.h",
"base/fake_video_renderer.cc",
"base/fake_video_renderer.h",
"base/test_utils.cc",
"base/test_utils.h",
"engine/fake_webrtc_call.cc",
"engine/fake_webrtc_call.h",
"engine/fake_webrtc_video_engine.cc",
"engine/fake_webrtc_video_engine.h",
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
]
}
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 (!build_with_chromium) {
rtc_media_unittests_resources = [
"../resources/media/captured-320x240-2s-48.frames",
"../resources/media/faces.1280x720_P420.yuv",
"../resources/media/faces_I400.jpg",
"../resources/media/faces_I411.jpg",
"../resources/media/faces_I420.jpg",
"../resources/media/faces_I422.jpg",
"../resources/media/faces_I444.jpg",
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
]
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 (is_ios) {
bundle_data("rtc_media_unittests_bundle_data") {
testonly = true
sources = rtc_media_unittests_resources
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
}
Fixed a couple of build-flag dependent tests of webrtcvoiceengine. The codecs expected by HasCorrectCodecs now depends which codecs were enabled by build flags. SendSideBweWithOverheadTest.MinAndMaxBitrate now expects different values for min bitrate depending on if we support 120 ms frames for Opus. BUG=b/35415435 Review-Url: https://codereview.webrtc.org/2691343008 Cr-Commit-Position: refs/heads/master@{#16643}
2017-02-16 05:37:06 -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
rtc_test("rtc_media_unittests") {
testonly = true
GN: Fix resource files for iOS test target In order to get resource files to be properly packaged into the .app for a unit test on iOS, the resource files needs to be listed as sources in a bundle_data target. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2292853002 Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 02:53:49 -07:00
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
defines = []
deps = [
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":codec",
Remove CreateMediaChannel calls from webrtc_video_engine_unittest Bug: webrtc:13931 Change-Id: I3d54741dffb337de9db80efa81b24396b96245f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309621 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40326}
2023-06-21 12:36:06 +00:00
":media_channel",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":media_constants",
Continue breakup of media/rtc_media_base Left in target are just .cc files with .h files used externally. Bug: webrtc:14775 Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 11:01:25 +00:00
":media_engine",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
":rid_description",
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
":rtc_audio_video",
":rtc_internal_video_codecs",
":rtc_media",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
":rtc_media_config",
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
":rtc_media_tests_utils",
Move h264_profile_level_id and vp9_profile to api/video_codecs This is a refactor to simplify a follow-up CL of adding SdpVideoFormat::IsSameCodec. The original files media/base/h264_profile_level_id.* and media/base/vp9_profile.h must be kept until downstream projects stop using them. Bug: chroimium:1187565 Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-19 09:09:26 +02:00
":rtc_sdp_video_format_utils",
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
":rtc_simulcast_encoder_adapter",
Split media_channel and its dependencies from the rtc_media_base target This helps in figuring out which dependencies exist, and gets closer to obeying the "one target per .cc file" rule. Test failures seem unrelated, so using No-Try. No-Try: true Bug: webrtc:14775 Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38910}
2022-12-16 09:54:57 +00:00
":rtp_utils",
":stream_params",
":turn_utils",
Remove remaining .cc files from rtc_media_base Also remove all dependencies on rtc_media_base except for a few that are suspected of being linker directives. Bug: webrtc:14775 Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41886}
2024-03-12 12:06:16 +00:00
":video_adapter",
":video_broadcaster",
":video_common",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:audio_options_api",
"../api:call_api",
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
"../api:create_simulcast_test_fixture_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 19/inf Convert most field trials used in PCLF tests. Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d Bug: webrtc:10335 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703 Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40909}
2023-10-10 15:21:29 +02:00
"../api:field_trials_view",
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
"../api:libjingle_peerconnection_api",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:make_ref_counted",
RtpSenderInterface::SetEncoderSelector This cl/ adds a way of setting an EncoderSelector on a specific RtpSenderInterface. This makes it possible to easily use different EncoderSelector on different streams within the same or different PeerConnections. The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor. Iff a EncoderSelector is set on the RtpSender, it will take precedence over the VideoEncoderFactory::GetEncoderSelector. Bug: webrtc:14122 Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:25:46 +02:00
"../api:mock_encoder_selector",
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
"../api:mock_video_bitrate_allocator",
"../api:mock_video_bitrate_allocator_factory",
"../api:mock_video_codec_factory",
"../api:mock_video_encoder",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../api:priority",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api:ref_count",
"../api:rtc_error",
"../api:rtp_headers",
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
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:simulcast_test_fixture_api",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../api:transport_api",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/audio:audio_processing",
Update VoiceEngine tests to use BuiltingAudioProcessingBuilder instead of AudioProcessingBuilder Bug: webrtc:369904700 Change-Id: I26115ef8d4a5f2997f8286eead07dc6cf28e9496 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367203 Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43339}
2024-10-31 13:57:50 +01:00
"../api/audio:builtin_audio_processing_builder",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/audio_codecs:audio_codecs_api",
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
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/crypto:options",
Update MediaEngine unittests to create Call using Environment Bug: webrtc:15656 Change-Id: I8016f03fd0640d218344f5a6ab53c4b0663690c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329081 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41260}
2023-11-28 17:30:07 +01:00
"../api/environment",
"../api/environment:environment_factory",
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
"../api/rtc_event_log",
"../api/task_queue",
"../api/test/video:function_video_factory",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/transport:bitrate_settings",
Fix dcsctp handling of dtls restart dtls_transport will when detecting a new fingerprint (e.g by usage of pranswer) signal DtlsTransportState::kNew. When this happen, the dtls crypto state is lost, and sctp should reconnect, srtp does this automatically in current code base. The existing behavior in dcsctp is that it will detect peer sending an init, and reconnect. But any messages sent between the dtls restart and the message arriving from the peer will be lost. This patch changes so that this case is gracefully handled by a) letting dcsctp_transport listen to dtls state this is big part of patch and involves changing the type of the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this into a separate patch... b) if a dtls restart happens, delete and restart socket. Testcase that fails before patch and works after is attached. Bonus: And include-what-you-use on patch Bug: b/375327137 Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43546}
2024-12-12 09:12:56 +01:00
"../api/transport:datagram_transport_interface",
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
"../api/transport:field_trial_based_config",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../api/transport/rtp:rtp_source",
SEA discards inactive encoders in implementation name Inactive encoders are included in the string when they are paused due to bitrate allocation being 0 for that simulcast layer. #rtc_ktlo Bug: webrtc:376804631 Change-Id: I4234b452b60fee58981907380df41962fda5bf40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43367}
2024-11-06 13:33:21 +00:00
"../api/units:data_rate",
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
"../api/units:time_delta",
WebRtcVideoEngineTest: use simulated time. Bug: chromium:1255737 Change-Id: I6036ae5af4b3f0e7bd04352b055935f501ecc52b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237341 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35316}
2021-11-05 15:02:03 +01:00
"../api/units:timestamp",
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
"../api/video:builtin_video_bitrate_allocator_factory",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../api/video:recordable_encoded_frame",
RtpEncodingParameters::request_resolution patch 4 This patch 1) modifies VideoAdapter to use requested_resolution instead on OnOutputFormatRequest, iff there are no active encoders that is not using requested_resolution (i.e all "old" encoder(s) are not active). 2) modifies VideoBroadcaster to not broadcast wants from encoders that are not active (iff there is an active encoder using requested_resolution). 3) fixes a bug in encoder_stream_factor in that the requested_resolution was not propagated to return value (must have been lost in merge?). Bug: webrtc:14451 Change-Id: I00e0907f0fe9329141ed169576fa46cdc5384886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278360 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38323}
2022-10-07 15:37:17 +02:00
"../api/video:resolution",
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
"../api/video:video_bitrate_allocation",
SEA discards inactive encoders in implementation name Inactive encoders are included in the string when they are paused due to bitrate allocation being 0 for that simulcast layer. #rtc_ktlo Bug: webrtc:376804631 Change-Id: I4234b452b60fee58981907380df41962fda5bf40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43367}
2024-11-06 13:33:21 +00:00
"../api/video:video_bitrate_allocator",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../api/video:video_bitrate_allocator_factory",
Create and initialize encoders only for active streams Bug: webrtc:12407 Change-Id: Id30fcb84dcbfffa30c7a34b15564ab5049cec96c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204066 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33141}
2021-02-02 18:23:40 +01:00
"../api/video:video_codec_constants",
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
"../api/video:video_frame",
"../api/video:video_rtp_headers",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../api/video_codecs:scalability_mode",
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
"../api/video_codecs:video_codecs_api",
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. Bug: webrtc:13573 Change-Id: I4eb5bb99bfde9c38c1d4072b933ef11a9ca32f53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299703 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39743}
2023-03-31 16:36:12 +02:00
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
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
"../audio",
"../call:call_interfaces",
Remove all references to codec-level transport-cc functions and flags. This seems to have no effect on tests, so it appears that these were not used after all. The goal is to make transport-cc a media-section-level attribute. Bug: webrtc:378698658 Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43411}
2024-11-18 07:18:35 +00:00
"../call:payload_type_picker",
Split up the call/video_stream_api target The split shows that some places don't need it at all. Most other places will depend on both send and receive stream targets. Bug: webrtc:373151158 Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43230}
2024-10-13 21:08:30 +00:00
"../call:video_receive_stream_api",
"../call:video_send_stream_api",
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
"../common_video",
"../modules/audio_device:mock_audio_device",
Fix crash when setting RTCRtpEncodingParameter.maxBitrate to a non-positive value. This will result in using a default value instead. The crash is caused by configuring audio send stream with a non-positive max_bitrate_bps and it expects a positive value (or special value -1). Also remove support for setting min bitrate since it is not in the spec and there are no tests. Bug: chromium:1377286 Change-Id: I19af29bff79eda5c0fbb1e528fced7976f5c2511 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284640 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38719}
2022-11-23 16:48:49 +01:00
"../modules/audio_mixer:audio_mixer_impl",
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
"../modules/audio_processing:mocks",
"../modules/rtp_rtcp",
Reduce usage of RtpHeaderParser::CreateForTest in favor of RtpPacket As a step to delete the legacy rtp packet parser. Bug: None Change-Id: I2aae86bc8847acd76cdd89007273a99f0298fdb9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221109 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34219}
2021-06-03 12:27:12 +02:00
"../modules/rtp_rtcp:rtp_rtcp_format",
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
"../modules/video_coding:simulcast_test_fixture_impl",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
Add SVC fallback. Fallback to a default value if the scalability mode is unset or not supported by the codec. The fallback logic is only enabled if the scalability mode is configured for any of the encodings for now (i.e. initial default values are not set). Bug: webrtc:11607 Change-Id: Ie632767b627a1dbbef71c59f9340573daf386c14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287600 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39074}
2023-01-11 15:50:58 +01:00
"../modules/video_coding/svc:scalability_mode_util",
Change DataChannelInit::priority to integer and forward to SCTP transport The new type PriorityValue is a strong 16-bit integer matching RFC 8831 requirements that can be built from a Priority enum. The value is now propagated and used by the SCTP transport, but enabling the feature still requires a field trial for now. Bug: webrtc:42225365 Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42695}
2024-07-26 16:16:41 +00:00
"../net/dcsctp/public:types",
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
"../p2p:p2p_test_utils",
Split rtc_base into multiple targets Keeping the headers to allow compatibility with current users that expect the headers to be in that target before they are also updated. Bug: webrtc:9838 Change-Id: I8b1e88850958e92c043686587a37791f01860220 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39031}
2023-01-04 12:01:09 +01:00
"../rtc_base:async_packet_socket",
Move byte_order out of rtc_base_approved Bug: webrtc:9838 Change-Id: Ieb2fbaad8831439ca04fc5249e295c8839f6890e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259309 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36633}
2022-04-19 17:18:36 +02:00
"../rtc_base:byte_order",
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
"../rtc_base:checks",
Use PayloadTypePicker in WebRtcVoiceEngine This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code. Bug: webrtc:360058654 Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42989}
2024-09-07 20:38:29 +00:00
"../rtc_base:copy_on_write_buffer",
"../rtc_base:dscp",
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
"../rtc_base:gunit_helpers",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../rtc_base:macromagic",
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
"../rtc_base:rtc_base_tests_utils",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../rtc_base:rtc_event",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../rtc_base:safe_conversions",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../rtc_base:socket",
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
"../rtc_base:stringutils",
"../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../rtc_base:timeutils",
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
"../rtc_base/experiments:min_video_bitrate_experiment",
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/sigslot",
Make VideoSendStream::UpdateActiveSimulcastLayers not block. UpdateActiveSimulcastLayers has been blocking WebRtcVideoChannel::SetSend which may be called quite frequently during negotiations. This CL changes UpdateActiveSimulcastLayers to not synchronize with the transport's task queue to wait for the changes to get applied. This synchronization is quite costly, but so too are other remaining things in VideoSendStream, so we should aim to get rid of the `thread_sync_event_` in VideoSendStream. Bug: webrtc:12840, webrtc:12854 Change-Id: Idb48d29b6b8382881c7c1e6f1d0f5e708dbca30f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221203 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34228}
2021-06-04 12:50:01 +02:00
"../system_wrappers:field_trial",
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
"../test:audio_codec_mocks",
"../test:fake_video_codecs",
"../test:field_trial",
"../test:rtp_test_utils",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 3/inf convert much of media/ (and the collateral) Bug: webrtc:10335 Change-Id: I04489dfe9622efe7f89e04aba3be6b3f60e77c91 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254520 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36177}
2022-03-10 15:21:28 +01:00
"../test:scoped_key_value_config",
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
"../test:test_main",
"../test:test_support",
"../test:video_test_common",
WebRtcVideoEngineTest: use simulated time. Bug: chromium:1255737 Change-Id: I6036ae5af4b3f0e7bd04352b055935f501ecc52b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237341 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35316}
2021-11-05 15:02:03 +01:00
"../test/time_controller",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"../video/config:encoder_config",
Move EncoderStreamFactory into own file This cl/ is a NOP refactoring, moving the EncoderStreamFactory from within webrtc_video_engine.cc into own file in video/. simulcast.cc is collateral. Bug: webrtc:14451 Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38224}
2022-09-27 16:53:04 +02:00
"../video/config:streams_config",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/algorithm:container",
Change cricket::Codec default id from 0 to -1 This allows detecting if it has been set reliably. 0 is a valid payload type. Bug: webrtc:360058654 Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43016}
2024-09-12 14:02:21 +00:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Mass removal of absl_deps in all BUILD.gn files Bug: webrtc:341803749 Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 11:21:53 +00:00
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
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
]
Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." This reverts commit f8c81ca4690aa3e470cc61633f512de383c308a8. Reason for revert: <Prepare to reland> Original change's description: > Revert "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate." > > This reverts commit 256733c78af655029cb04afae2c404afb41ea685. > > Reason for revert: <breaks downstream> > > Original change's description: > > Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate. > > > > Bug: webrtc:13573 > > Change-Id: Iae649e7922a67f70c53d33f3b87ea62bb6a3490c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262812 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36940} > > Bug: webrtc:13573 > Change-Id: I3341b6b96a56de63058c38943611b8c1629294ce > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262941 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36942} Bug: webrtc:13573 Change-Id: Iaf1222c58a18f378df20e4f83262b9a9da491712 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262943 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36985}
2022-05-24 10:32:32 +02:00
if (enable_libaom) {
defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
}
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
sources = [
Add comparators unittest, and abandon MatchesForSdp Use the same code in PayloadTypePicker as in Codec.Matches() Bug: webrtc:360058654 Change-Id: I549ed24860648cfdb6a173a19773daf01db827b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365102 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43217}
2024-10-10 12:40:57 +00:00
"base/codec_comparators_unittest.cc",
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
"base/codec_unittest.cc",
"base/media_engine_unittest.cc",
"base/rtp_utils_unittest.cc",
Move h264_profile_level_id and vp9_profile to api/video_codecs This is a refactor to simplify a follow-up CL of adding SdpVideoFormat::IsSameCodec. The original files media/base/h264_profile_level_id.* and media/base/vp9_profile.h must be kept until downstream projects stop using them. Bug: chroimium:1187565 Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33782}
2021-04-19 09:09:26 +02:00
"base/sdp_video_format_utils_unittest.cc",
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
"base/stream_params_unittest.cc",
"base/turn_utils_unittest.cc",
"base/video_adapter_unittest.cc",
"base/video_broadcaster_unittest.cc",
"base/video_common_unittest.cc",
"engine/internal_decoder_factory_unittest.cc",
Reland "Handle scalability mode in QueryCodecSupport" This reverts commit 74281bed5350af9c15f83e0b1aec5c5921dbf76f. Reason for revert: Fixed unit test by removing VP9 profile 2 from encoder factory unit test since this is platform dependent. Original change's description: > Revert "Handle scalability mode in QueryCodecSupport" > > This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5. > > Reason for revert: Speculative revert. Breaks upstream project http://b/200009579 > > Original change's description: > > Handle scalability mode in QueryCodecSupport > > > > All valid scalability modes should be supported by the builtin > > software decoder/encoder. > > > > Bug: chromium:1187565 > > Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34998} > > TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1187565 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35001} Bug: chromium:1187565 Change-Id: I598a2a530b8fea22997bbb5910eb3b864d1e28a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232021 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35003}
2021-09-15 10:56:04 +00:00
"engine/internal_encoder_factory_unittest.cc",
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
"engine/simulcast_encoder_adapter_unittest.cc",
"engine/webrtc_media_engine_unittest.cc",
"engine/webrtc_video_engine_unittest.cc",
]
Relanding: Fix data channel message integrity violation Patch originally submitted by Lennart Grahl: https://webrtc-review.googlesource.com/c/src/+/177527 SCTP message chunks and notifications are being delivered interleaved. However, the way the code was structured previously, a notification would interrupt reassembly of a message chunk and hand out the partial message, thereby violating message integrity. This patch separates the handling of notifications and reassembly of messages. Additional changes: - Remove illegal cast from non-validated u32 to enum (PPID) - Drop partial messages if the SID has been changed but EOR not yet received instead of delivering them. (This should never happen anyway.) - Don't treat TSN as timestamp (wat) - Replace "usrsctplib/usrsctp.h" with <usrsctp.h>, allowing a hack to be removed from media/BUILD.gn Bug: webrtc:11708 Change-Id: I29733b03f67a3d840104b8608a7f0083466c2d0f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178469 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31641}
2020-07-03 16:37:57 -07:00
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
# TODO(kthelgason): Reenable this test on iOS.
# See bugs.webrtc.org/5569
if (!is_ios) {
sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
}
if (rtc_opus_support_120ms_ptime) {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
} else {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
}
data = rtc_media_unittests_resources
if (is_android) {
shard_timeout = 900
}
if (is_ios) {
deps += [ ":rtc_media_unittests_bundle_data" ]
}
sctp: Fix data channel closing sequence When an SCTP stream is closing, a stream reset needs to be sent from both ends. The remote was not sending a stream reset and quickly opening another stream with the same StreamID could cause SCTP errors. Bug: webrtc:13994 Change-Id: I3abc74ddc88b3fcf7e6495d76e7d77f52280b5d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260922 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36773}
2022-05-03 00:24:15 +02:00
if (rtc_build_dcsctp) {
sources += [ "sctp/dcsctp_transport_unittest.cc" ]
deps += [
":rtc_data_dcsctp_transport",
"../net/dcsctp/public:factory",
"../net/dcsctp/public:mocks",
"../net/dcsctp/public:socket",
]
}
Relanding: Fix data channel message integrity violation Patch originally submitted by Lennart Grahl: https://webrtc-review.googlesource.com/c/src/+/177527 SCTP message chunks and notifications are being delivered interleaved. However, the way the code was structured previously, a notification would interrupt reassembly of a message chunk and hand out the partial message, thereby violating message integrity. This patch separates the handling of notifications and reassembly of messages. Additional changes: - Remove illegal cast from non-validated u32 to enum (PPID) - Drop partial messages if the SID has been changed but EOR not yet received instead of delivering them. (This should never happen anyway.) - Don't treat TSN as timestamp (wat) - Replace "usrsctplib/usrsctp.h" with <usrsctp.h>, allowing a hack to be removed from media/BUILD.gn Bug: webrtc:11708 Change-Id: I29733b03f67a3d840104b8608a7f0083466c2d0f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178469 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31641}
2020-07-03 16:37:57 -07:00
}
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
2016-06-12 22:12:01 -07:00
}
}
Reference in New Issue Copy Permalink