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Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("//build/config/linux/pkg_config.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 = [
"../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",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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 = []
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 = [
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",
[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",
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",
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",
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",
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",
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:webrtc_key_value_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",
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",
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",
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",
WebRtcVideoEngine: Enable encoded frame sink. This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames can flow to sinks installed in VideoTrackSourceInterface. Bug: chromium:1013590 Change-Id: I136132c210e5811547f2522ddc371d0acac90664 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161093 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30001}
2019-12-04 10:58:17 +01:00
"../call:video_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",
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",
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_processing:audio_processing_statistics",
[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",
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",
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",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_base_approved",
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:rtc_base_approved",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
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",
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",
Remove lock from MediaChannel Pending messages on network thread for MediaChannel, will be dropped when the MediaChannel object is deleted (without blocking). Remove MessageHandler inheritance from Channel since Post-ing to the network thread has been removed from there. Copy/pasted code for SendRtp/SendRtcp in WebRtcVideoChannel and WebRtcVoiceMediaChannel consolidated in MediaChannel. Bug: webrtc:11993 Change-Id: I05320eb7f86b98adba50ca5eb8b76b78f4111263 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217720 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33955}
2021-05-07 18:02:53 +02:00
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
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",
Add option to make first scale factor depend on input resolution. Scale factors are 3/4, 2/3, 3/4, 2/3, ... Adds possibly to start with: - 2/3 (if width/height multiple of 3) - 2/3, 2/3 (if width/height multiple of 9) Bug: none Change-Id: Idbeddfec4baea893c240bbb897d01ac1cff3b435 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157105 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29538}
2019-10-18 15:03:13 +02:00
"../system_wrappers:field_trial",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
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
]
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/adapted_video_track_source.cc",
"base/adapted_video_track_source.h",
"base/audio_source.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
"base/codec.cc",
"base/codec.h",
Propagate base minimum delay from video jitter buffer to webrtc/api. On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface, GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes the same concept. Bug: webrtc:10287 Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957 Reviewed-on: https://webrtc-review.googlesource.com/c/123482 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26877}
2019-02-27 15:32:48 +01:00
"base/delayable.h",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"base/media_channel.cc",
"base/media_channel.h",
"base/media_constants.cc",
"base/media_constants.h",
"base/media_engine.cc",
"base/media_engine.h",
"base/rid_description.cc",
"base/rid_description.h",
"base/rtp_utils.cc",
"base/rtp_utils.h",
"base/stream_params.cc",
"base/stream_params.h",
"base/turn_utils.cc",
"base/turn_utils.h",
"base/video_adapter.cc",
"base/video_adapter.h",
"base/video_broadcaster.cc",
"base/video_broadcaster.h",
"base/video_common.cc",
"base/video_common.h",
"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
]
}
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 = [
Signal requested resolution alignment requirements from sinks to sources. Bug: webrtc:11218 Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 09:47:11 +01:00
":rtc_media_base",
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",
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",
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",
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",
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",
WebRtcVideoEngine: Enable encoded frame sink. This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded frames can flow to sinks installed in VideoTrackSourceInterface. Bug: chromium:1013590 Change-Id: I136132c210e5811547f2522ddc371d0acac90664 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161093 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30001}
2019-12-04 10:58:17 +01:00
"../call:video_stream_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",
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",
"../rtc_base:rtc_base_approved",
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",
"../system_wrappers:field_trial",
]
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
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/types:optional",
]
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_encoder_simulcast_proxy") {
Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy Use the new class internally where appropriate too. The objective is to rename it, but due to some external dependency, it is better to copy, update dependencies and remove. Bug: webrtc:10069 Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3 Reviewed-on: https://webrtc-review.googlesource.com/c/113265 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:38:24 +01:00
visibility = [ "*" ]
defines = []
libs = []
sources = [
"engine/encoder_simulcast_proxy.cc",
"engine/encoder_simulcast_proxy.h",
]
deps = [
":rtc_simulcast_encoder_adapter",
"../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",
Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy Use the new class internally where appropriate too. The objective is to rename it, but due to some external dependency, it is better to copy, update dependencies and remove. Bug: webrtc:10069 Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3 Reviewed-on: https://webrtc-review.googlesource.com/c/113265 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:38:24 +01:00
"../api/video_codecs:video_codecs_api",
"../modules/video_coding:video_codec_interface",
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",
Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy Use the new class internally where appropriate too. The objective is to rename it, but due to some external dependency, it is better to copy, update dependencies and remove. Bug: webrtc:10069 Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3 Reviewed-on: https://webrtc-review.googlesource.com/c/113265 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:38:24 +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_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 = [
Copy VP8EncoderSimulcastProxy to EncoderSimulcastProxy Use the new class internally where appropriate too. The objective is to rename it, but due to some external dependency, it is better to copy, update dependencies and remove. Bug: webrtc:10069 Change-Id: I8477ce5a2982933db27513cc9509f51558dafaf3 Reviewed-on: https://webrtc-review.googlesource.com/c/113265 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25920}
2018-12-06 13:38:24 +01:00
":rtc_encoder_simulcast_proxy",
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_base",
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",
[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 "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",
"../call:call_interfaces",
"../call:video_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
"../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",
Reland "Rename stereo video codec to multiplex" This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174. Original change's description: > Rename stereo video codec to multiplex > > This CL only does the rename from"stereo" to multiplex". With this we have a > better name that doesn't clash with audio's usage of stereo. > > Bug: webrtc:7671 > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652 > Reviewed-on: https://webrtc-review.googlesource.com/43242 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21769} TBR=niklas.enbom@webrtc.org Bug: webrtc:7671 Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511 Reviewed-on: https://webrtc-review.googlesource.com/45820 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21794}
2018-01-25 13:01:09 -08:00
"../modules/video_coding:webrtc_multiplex",
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_vp8",
"../modules/video_coding:webrtc_vp9",
Add Av1 Decoder wrapper behind a build flag Bug: webrtc:11404 Change-Id: I090ffd173d667e8845de1b986af462516b7c76e6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169452 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30757}
2020-03-11 10:45:57 +01:00
"../modules/video_coding/codecs/av1:libaom_av1_decoder",
Make libaom_av1_encoder always build the libaom encoder. Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance. Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`. Bug: webrtc:13573 Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35763}
2022-01-18 18:47:52 +01:00
"../modules/video_coding/codecs/av1:libaom_av1_encoder_if_supported",
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",
"../rtc_base:rtc_base_approved",
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 "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 "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" ]
}
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
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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",
"engine/multiplex_codec_factory.cc",
"engine/multiplex_codec_factory.h",
# TODO(bugs.webrtc.org/7925): stop exporting this header once downstream
# targets depend on :rtc_encoder_simulcast_proxy directly.
"engine/encoder_simulcast_proxy.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 = [
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_base",
"../api:call_api",
"../api:libjingle_peerconnection_api",
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",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
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",
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",
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",
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",
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",
"../api/transport:webrtc_key_value_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",
[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",
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",
"../api/video_codecs:video_codecs_api",
"../call",
"../call:call_interfaces",
"../call:video_stream_api",
"../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: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",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../modules/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",
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",
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",
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",
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",
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
"../rtc_base:ignore_wundef",
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:rtc_task_queue",
"../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",
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",
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",
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",
Call MediaChannel::OnPacketReceived on the network thread. Functionality wise, there should be no change with this CL, aside from updating tests to anticipate OnPacketReceived to handle the packet asynchronously (as already was the case via BaseChannel). This only removes the network->worker hop out of the BaseChannel class into the WebRTC MediaChannel implementations. However, it updates the interface contract between BaseChannel and MediaChannel to align with how we want things to work down the line, i.e. avoid hopping to the worker thread for every rtp packet. The following steps will be to update the video and voice channel classes to call Call::DeliverPacket on the network thread and only handle unsignalled SSRCs on the worker (exception case). Bug: webrtc:11993 Change-Id: If0540874444565dc93773aee89d862f3bfc9c502 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202242 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33040}
2021-01-18 10:49:05 +01:00
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
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",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
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/null_webrtc_video_engine.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
"engine/payload_type_mapper.cc",
"engine/payload_type_mapper.h",
Enable building WebRTC without built-in software codecs This CL adds a GN build flag to include builtin software codecs (enabled by default). When setting the flag to false, libvpx can also be excluded. The benefit is that the resulting binary is smaller. Replaces https://webrtc-review.googlesource.com/c/src/+/29203 Bug: webrtc:7925 Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10 Reviewed-on: https://webrtc-review.googlesource.com/36340 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-30 10:32:13 +01:00
"engine/simulcast.cc",
"engine/simulcast.h",
Improve handling of packets with unknown ssrc. Add a feature (gated by field trial) that stores packets with unknown ssrc in a circular buffer and replays them once a receive stream with matching ssrc is created. This improves situation where media is incoming but signaling or SetFrameDecryptor is slow. BUG=webrtc:10405 Change-Id: I7c7b2f4bd96c942c09e96db0cdae4ce5efef2541 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127543 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27159}
2019-03-18 10:59:40 +01:00
"engine/unhandled_packets_buffer.cc",
"engine/unhandled_packets_buffer.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
}
Introduce media engine factory where TaskQueueFactory dependency can be set. For new factory function use same style as PeerConnectionFactory does: insteat of multiple parameters pass struct where some parameters might be not set. Bug: webrtc:10284 Change-Id: Ic54813e3afa3f873295409d2f7e2347c69f76988 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131952 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27556}
2019-04-10 14:10:10 +02:00
# Heavy but optional helper for unittests and webrtc users who prefer to use
# defaults factories or do not worry about extra dependencies and binary size.
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_engine_defaults") {
Introduce media engine factory where TaskQueueFactory dependency can be set. For new factory function use same style as PeerConnectionFactory does: insteat of multiple parameters pass struct where some parameters might be not set. Bug: webrtc:10284 Change-Id: Ic54813e3afa3f873295409d2f7e2347c69f76988 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131952 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27556}
2019-04-10 14:10:10 +02:00
visibility = [ "*" ]
allow_poison = [
"audio_codecs",
Poison default task queue factory Users of webrtc generally should be able to choose own task queue implementation. Poison avoids accidental dependency of a low level component on the default implementation Android and ios apis are still de-facto forced to use the default implementation. Bug: webrtc:10284 Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28524}
2019-07-10 12:44:43 +02:00
"default_task_queue",
Introduce media engine factory where TaskQueueFactory dependency can be set. For new factory function use same style as PeerConnectionFactory does: insteat of multiple parameters pass struct where some parameters might be not set. Bug: webrtc:10284 Change-Id: Ic54813e3afa3f873295409d2f7e2347c69f76988 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131952 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27556}
2019-04-10 14:10:10 +02:00
"software_video_codecs",
]
sources = [
"engine/webrtc_media_engine_defaults.cc",
"engine/webrtc_media_engine_defaults.h",
]
deps = [
":rtc_audio_video",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/task_queue:default_task_queue_factory",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../modules/audio_processing:api",
"../rtc_base:checks",
Export more symbols to fix Chromecast component build When building certain Chromecast build flavors in component build mode, there are some link errors due to symbols not being exported. This CL fixes those issues. TBR: kwiberg@webrtc.org Bug: None Change-Id: I408f0a84b8ac4610cc6b5aa6ff58248ea82c9c66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161148 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29992}
2019-12-02 10:26:34 -08:00
"../rtc_base/system:rtc_export",
Introduce media engine factory where TaskQueueFactory dependency can be set. For new factory function use same style as PeerConnectionFactory does: insteat of multiple parameters pass struct where some parameters might be not set. Bug: webrtc:10284 Change-Id: Ic54813e3afa3f873295409d2f7e2347c69f76988 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131952 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27556}
2019-04-10 14:10:10 +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 = [
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",
sctp: Reorganize build targets Bug: webrtc:12614 Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33745}
2021-04-15 15:02:56 +02:00
"../media:rtc_media_base",
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
"../p2p:rtc_p2p",
"../rtc_base:rtc_base_approved",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
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",
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 = [
":rtc_data_sctp_transport_internal",
"../api:array_view",
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",
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
"../media:rtc_media_base",
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",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
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",
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",
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
"../rtc_base/third_party/sigslot:sigslot",
"../system_wrappers",
]
absl_deps += [
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
}
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
if (rtc_build_usrsctp) {
rtc_library("rtc_data_usrsctp_transport") {
defines = [
# "SCTP_DEBUG" # Uncomment for SCTP debugging.
]
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
sources = [
sctp: Rename SctpTransport to UsrSctpTransport The rename ensures we don't confuse this implementation with the new one based on the new dcSCTP library. Bug: webrtc:12614 No-Presubmit: True Change-Id: Ida08659bbea9c98aba8247d4368799ff7dd18729 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214482 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33688}
2021-04-08 15:06:09 +02:00
"sctp/usrsctp_transport.cc",
"sctp/usrsctp_transport.h",
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: 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_data_sctp_transport_internal",
"../media:rtc_media_base",
"../p2p:rtc_p2p",
"../rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
"../rtc_base/third_party/sigslot:sigslot",
Reland "Make cricket::SctpTransportInternalFactory injectable through PeerConnectionFactory Deps" This is to allow testing without using the singleton sctp library. cricket::SctpTransportInternalFactory is renamed to webrtc::SctpTransportFactoryInterface and moved to the API folder to follow the API structure. Tests can use test/pc/sctp/fake_sctp_transport.h to inject a faked data channel implementation. patch 1 contain the original cl. patch 2 modifications Bug: none Change-Id: Ic088da3eb7d9aada79e6d601dbf2d1aa2be777f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182840 Reviewed-by: Taylor <deadbeef@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32024}
2020-08-28 09:15:15 +02:00
"//third_party/usrsctp",
]
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
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/types:optional",
]
}
}
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",
"../api/transport:sctp_transport_factory_interface",
"../rtc_base:threading",
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/experiments:field_trial_parser",
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/system:unused",
]
if (rtc_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
assert(rtc_build_dcsctp || rtc_build_usrsctp,
"An SCTP backend is required to enable SCTP")
}
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
}
if (rtc_build_usrsctp) {
defines += [ "WEBRTC_HAVE_USRSCTP" ]
deps += [ ":rtc_data_usrsctp_transport" ]
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 = [
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",
":rtc_media_base",
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",
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",
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",
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",
[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",
Move video send/receive stream headers to webrtc/call. Moved the headers video_receive_stream.h and video_send_stream.h from webrtc/ into webrtc/call/ as part of the Slim and Modular work. The GN target webrtc:video_stream_api has moved to webrtc/call:video_stream_api. There are headers left in webrtc/ with the same name including the moved headers in webrtc/call/ for not breaking external projects depending on WebRTC. At the same time, some minor cleanup is done: Non-pure-virtual functions declared in the two affected headers now have definitions in the same target. After making this change, our 'chromium-style' plugin detected some style violations that have now been fixed: non-inlined constructors and destructors have been added to a number of classes, both inside the GN target of the two affected headers, and in other targets. BUG=webrtc:8107 Review-Url: https://codereview.webrtc.org/3000253002 Cr-Commit-Position: refs/heads/master@{#19448}
2017-08-22 05:43:23 -07:00
"../call:video_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",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
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",
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
"../p2p:rtc_p2p",
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",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../rtc_base:checks",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:gunit_helpers",
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:rtc_base_approved",
Detangling target dependencies in rtc_base_approved. The eventual goal is to allow PlatformThread to use SequencedTaskChecker, but getting to that point will require some more detangling. Here are (roughly) the steps taken in this CL: * Make constructormagic a separate target. * Move atomicops and arraysize to separate targets * Move platform_thread_types to a separate target * Move criticalsection to a separate target * Move thread_checker to separate target * Make sequenced_task_checker not depend on base_approved * Move ptr_util to a separate target * Move scoped_ptr to ptr_util * Make rtc_task_queue_api not depend on base_approved * Make sequenced_task_checker depend on rtc_task_queue_api * Move rtc::Event to its own target * Move basictypes.h to constructormagic * Move format_macros and stringize_macros into constructormagic * Rename constructormagic target to... macromagic * Move stringencode to stringutils * New target for safe_conversions * Move timeutils to a new target. * Move logging to a new target. * Move platform_thread to a new target. * Make refcount a new target (refcount, refcountedobject, refcounter). * Remove rtc_base_approved from deps of TQ * Remove a circular dependency between event tracer and platform thread. Further steps will probably be to factor TaskQueue::Current() to not be a part of the TaskQueue class itself and have it declared+implemented in a target that's lower level than TQ itself. SequencedTaskChecker can then depend on that target and avoid the TQ dependency. Once we're there, PlatformThread will be able to depend on SequencedTaskChecker. Attempted but eventually removed from this CL: * Make TQ a part of rtc_base_approved * Remove direct dependencies on sequenced_task_checker. * Profit. A few include-what-you-use updates along the way. Fix a few targets that were depending on rtc_task_queue_api Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9 Bug: webrtc:8957 Reviewed-on: https://webrtc-review.googlesource.com/58480 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:12:48 +01:00
"../rtc_base:rtc_task_queue",
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",
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",
"../test:test_support",
"//testing/gtest",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Use Abseil container algorithms in 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",
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",
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 = [
":rtc_audio_video",
":rtc_encoder_simulcast_proxy",
":rtc_internal_video_codecs",
":rtc_media",
":rtc_media_base",
":rtc_media_engine_defaults",
":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",
"../api:create_simulcast_test_fixture_api",
"../api:libjingle_peerconnection_api",
"../api:mock_video_bitrate_allocator",
"../api:mock_video_bitrate_allocator_factory",
"../api:mock_video_codec_factory",
"../api:mock_video_encoder",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:simulcast_test_fixture_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/rtc_event_log",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/transport:field_trial_based_config",
"../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",
"../api/video:video_bitrate_allocation",
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",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../audio",
"../call:call_interfaces",
"../common_video",
"../modules/audio_device:mock_audio_device",
"../modules/audio_processing",
"../modules/audio_processing:api",
"../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",
"../modules/video_coding/codecs/av1:libaom_av1_decoder",
Make libaom_av1_encoder always build the libaom encoder. Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance. Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`. Bug: webrtc:13573 Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35763}
2022-01-18 18:47:52 +01:00
"../modules/video_coding/codecs/av1:libaom_av1_encoder_if_supported",
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",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../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",
"../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",
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
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
sources = [
"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/encoder_simulcast_proxy_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/multiplex_codec_factory_unittest.cc",
"engine/null_webrtc_video_engine_unittest.cc",
"engine/payload_type_mapper_unittest.cc",
"engine/simulcast_encoder_adapter_unittest.cc",
"engine/simulcast_unittest.cc",
"engine/unhandled_packets_buffer_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" ]
}
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
if (rtc_build_usrsctp) {
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 += [
sctp: Rename SctpTransport to UsrSctpTransport The rename ensures we don't confuse this implementation with the new one based on the new dcSCTP library. Bug: webrtc:12614 No-Presubmit: True Change-Id: Ida08659bbea9c98aba8247d4368799ff7dd18729 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214482 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33688}
2021-04-08 15:06:09 +02:00
"sctp/usrsctp_transport_reliability_unittest.cc",
"sctp/usrsctp_transport_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
]
Delete AsyncInvoker usage from SctpDataSender Bug: webrtc:12339 Change-Id: I018137103c45e380c89cd4bdeefd434c1d4252e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212862 Reviewed-by: Taylor <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33556}
2021-03-23 12:29:07 +01:00
deps += [
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_data_sctp_transport_internal",
":rtc_data_usrsctp_transport",
Delete AsyncInvoker usage from SimulatedPacketTransport Bug: webrtc:12339 Change-Id: Ic293f9c8791ec24025f9eac39cbc4fcf2583d3ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212867 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33741}
2021-04-15 11:42:50 +02:00
"../rtc_base:rtc_event",
Delete AsyncInvoker usage from SctpDataSender Bug: webrtc:12339 Change-Id: I018137103c45e380c89cd4bdeefd434c1d4252e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212862 Reviewed-by: Taylor <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33556}
2021-03-23 12:29:07 +01:00
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
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
"//third_party/usrsctp",
Delete AsyncInvoker usage from SctpDataSender Bug: webrtc:12339 Change-Id: I018137103c45e380c89cd4bdeefd434c1d4252e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212862 Reviewed-by: Taylor <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33556}
2021-03-23 12:29:07 +01:00
]
Remove from chromium build targets that are not compatible with it. We need to be able build chromium with rtc_include_tests = true. It reveals a lot of targets that are not compatible with chromium but aren't marked so. `rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated. Bug: webrtc:12404 Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33124}
2021-02-01 09:56:37 +00:00
}
if (rtc_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) {
deps += [ "//testing/android/native_test:native_test_support" ]
shard_timeout = 900
}
if (is_ios) {
deps += [ ":rtc_media_unittests_bundle_data" ]
}
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
}
}
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