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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
}
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_h264_profile_id") {
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 Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) Reason for revert: Relanding the orginal CL. The breakage would be a flakey build. Original issue's description: > Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) > > Reason for revert: > The Android 32 (more config) bot is broken. > > Original issue's description: > > Try to fix the binary size increase issue on Chromium. > > > > The target common_video used to depend on rtc_media_base which introduces > > the dependency on p2p. This probably causes the binary size increase on Win > > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly. > > > > BUG=chromium:734631 > > > > Review-Url: https://codereview.webrtc.org/2945233002 > > Cr-Commit-Position: refs/heads/master@{#18693} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9ed16097375fb8d9b45623c58d9086d33e503760 > > TBR=kjellander@webrtc.org,deadbeef@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=chromium:734631 > > Review-Url: https://codereview.webrtc.org/2949953003 > Cr-Commit-Position: refs/heads/master@{#18694} > Committed: https://chromium.googlesource.com/external/webrtc/+/c2e208a9249452590fa282ef5aba43e480bc5794 TBR=kjellander@webrtc.org,deadbeef@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:734631 Review-Url: https://codereview.webrtc.org/2949883003 Cr-Commit-Position: refs/heads/master@{#18695}
2017-06-21 01:02:59 -07:00
sources = [
"base/h264_profile_level_id.cc",
"base/h264_profile_level_id.h",
]
deps = [
"..:webrtc_common",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
Remove CodecInst pt.2 The following APIs on AudioCodingModule are deprecated with this CL: static int NumberOfCodecs(); static int Codec(int, CodecInst*); static int Codec(const char*, CodecInst*, int, size_t); static int Codec(const char*, int, size_t); absl::optional<CodecInst> SendCodec() const; bool RegisterReceiveCodec(int, const SdpAudioFormat&); int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&); int UnregisterReceiveCodec(uint8_t); int32_t ReceiveCodec(CodecInst*); absl::optional<SdpAudioFormat> ReceiveFormat(); As well as this method on RtpRtcp module: int32_t RegisterSendPayload(const CodecInst&); Bug: webrtc:7626 Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce Reviewed-on: https://webrtc-review.googlesource.com/c/113816 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26025}
2018-12-11 12:22:10 +01:00
"../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../rtc_base:rtc_base_approved",
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",
Replace rtc::Optional with absl::optional in media, ortc, p2p This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script with parameters 'media ortc p2p': find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I19167714af7cc1436d34cfcba6c8b3718d8e677b Reviewed-on: https://webrtc-review.googlesource.com/83731 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23638}
2018-06-15 15:58:38 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) Reason for revert: Relanding the orginal CL. The breakage would be a flakey build. Original issue's description: > Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) > > Reason for revert: > The Android 32 (more config) bot is broken. > > Original issue's description: > > Try to fix the binary size increase issue on Chromium. > > > > The target common_video used to depend on rtc_media_base which introduces > > the dependency on p2p. This probably causes the binary size increase on Win > > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly. > > > > BUG=chromium:734631 > > > > Review-Url: https://codereview.webrtc.org/2945233002 > > Cr-Commit-Position: refs/heads/master@{#18693} > > Committed: https://chromium.googlesource.com/external/webrtc/+/9ed16097375fb8d9b45623c58d9086d33e503760 > > TBR=kjellander@webrtc.org,deadbeef@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=chromium:734631 > > Review-Url: https://codereview.webrtc.org/2949953003 > Cr-Commit-Position: refs/heads/master@{#18694} > Committed: https://chromium.googlesource.com/external/webrtc/+/c2e208a9249452590fa282ef5aba43e480bc5794 TBR=kjellander@webrtc.org,deadbeef@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:734631 Review-Url: https://codereview.webrtc.org/2949883003 Cr-Commit-Position: refs/heads/master@{#18695}
2017-06-21 01:02:59 -07:00
]
}
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 = [ "*" ]
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/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
]
}
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_vp9_profile") {
Add visibility for rtc_vp9_profile This is necessary to access profiles from Chrome side. Bug: webrtc:7925 Change-Id: I27d187afb56da715caf9f2ac8a6942778853542c Reviewed-on: https://webrtc-review.googlesource.com/c/113100 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25925}
2018-12-04 11:00:52 -05:00
visibility = [ "*" ]
Add VP9 profile negotiation to SDP This CL adds VP9 profile information in SDP. It adds the necessary fields and enums to codec containers. Additional profiles will be followed. Bug: webrtc:9376 Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54 Reviewed-on: https://webrtc-review.googlesource.com/81960 Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23810}
2018-06-28 10:59:02 -07:00
sources = [
"base/vp9_profile.cc",
"base/vp9_profile.h",
]
deps = [
"..:webrtc_common",
"../api/video_codecs:video_codecs_api",
"../rtc_base:rtc_base_approved",
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",
Add VP9 profile negotiation to SDP This CL adds VP9 profile information in SDP. It adds the necessary fields and enums to codec containers. Additional profiles will be followed. Bug: webrtc:9376 Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54 Reviewed-on: https://webrtc-review.googlesource.com/81960 Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23810}
2018-06-28 10:59:02 -07:00
"//third_party/abseil-cpp/absl/types:optional",
]
}
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_h264_profile_id",
":rtc_media_config",
":rtc_vp9_profile",
"..:webrtc_common",
[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",
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",
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",
Move MediaTransportInterface out of the libjingle_peerconnection_api target And move related files into api/transport/ and api/transport/media/. The moved files are unchanged, except that congestion_control_interface.h and datagram_transport_interface.h no longer include media_transport_interface.h, instead, they forward declare the few MediaTransport* types they reference. Bug: webrtc:8733 Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-12 13:59:36 +02:00
"../api/transport/media:media_transport_interface",
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 15:16:49 +02:00
"../api/transport/rtp:rtp_source",
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",
"../api/video:video_frame_i420",
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",
"../common_video",
"../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",
Reland "New build target p2p:stun_types" This is a reland of 5b4fcb5bf69218c2f42ca2b0cada6c15f2f638e9 Original change's description: > New build target p2p:stun_types > > The media:rtc_media_base target needs definitions of various > stun-related types and constant. With this new smaller target, it no > longer needs to depend on all of p2p. > > Bug: webrtc:8733 > Change-Id: I05910b6915f6d2c96e8f52a017adbc7eb693dca8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150945 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29036} Tbr: steveanton@webrtc.org Bug: webrtc:8733 Change-Id: I1847007ecf29e0e6a27f559b92df632a1cd69280 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151880 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29092}
2019-09-02 13:09:58 +02:00
"../p2p:stun_types",
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",
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",
Replacing SequencedTaskChecker with SequenceChecker. Bug: webrtc:9883 Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 13:44:04 +02:00
"../rtc_base/synchronization:sequence_checker",
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",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/sigslot",
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",
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_data_engine.cc",
"base/rtp_data_engine.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_constants") {
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
defines = []
libs = []
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
deps = []
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
sources = [
Move const variables initialization to constants.cc Bug: None Change-Id: I73f0d343e242e959879c5396852270c78c17fc87 Reviewed-on: https://webrtc-review.googlesource.com/22420 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20828}
2017-11-20 17:48:13 +05:30
"engine/constants.cc",
Fix Gn untracked headers in webrtc/media BUG=webrtc:7645 Review-Url: https://codereview.webrtc.org/3013063002 Cr-Original-Commit-Position: refs/heads/master@{#19824} Review-Url: https://codereview.webrtc.org/3018473002 Cr-Commit-Position: refs/heads/master@{#19844}
2017-09-14 13:53:02 -07:00
"engine/constants.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_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 = [
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",
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",
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:video_frame_i420",
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",
"../modules/video_coding:video_codec_interface",
"../modules/video_coding:video_coding_utility",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
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",
Replacing SequencedTaskChecker with SequenceChecker. Bug: webrtc:9883 Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 13:44:04 +02:00
"../rtc_base/synchronization:sequence_checker",
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",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/libyuv",
]
}
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 = [
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_constants",
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",
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_h264_profile_id",
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
"../:webrtc_common",
"../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",
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
"../modules:module_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",
"../rtc_base:checks",
Mark functions using old factory classes as deprecated. The flag rtc_use_builtin_sw_codecs will be removed in a later CL and this marks usage of the various entry points using the old video factory API as deprecated. Bug: webrtc:7925, webrtc:10044 Change-Id: I5c75516a41b0666e77539c028808cc2b173ed4bd Reviewed-on: https://webrtc-review.googlesource.com/c/113061 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25970}
2018-12-11 15:44:54 +01:00
"../rtc_base:deprecation",
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: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 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 "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42 Original change's description: > Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase > > Bug: None > Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad > Reviewed-on: https://webrtc-review.googlesource.com/c/107303 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25312} Tbr: kwiberg@webrtc.org Bug: None Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc Reviewed-on: https://webrtc-review.googlesource.com/c/107716 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-23 10:07:25 +02:00
"//third_party/abseil-cpp/absl/strings",
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
]
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_constants",
":rtc_media_base",
"..:webrtc_common",
"../api:call_api",
"../api:libjingle_peerconnection_api",
New target for api/rtp_parameters.h and api/media_types.h. The new target does not depend on libjingle_peerconnection_api, and to do this, the named "audio" and "video" string literals had to be moved from media_stream_interface.cc to media_types.cc. In this cl, the dependency on libjingle_peerconnection_api can be dropped from a few targets. No-Presubmit: True Bug: webrtc:8733 Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28998}
2019-08-26 10:45:28 +02:00
"../api:rtp_parameters",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../api:scoped_refptr",
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",
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",
Move MediaTransportInterface out of the libjingle_peerconnection_api target And move related files into api/transport/ and api/transport/media/. The moved files are unchanged, except that congestion_control_interface.h and datagram_transport_interface.h no longer include media_transport_interface.h, instead, they forward declare the few MediaTransport* types they reference. Bug: webrtc:8733 Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-12 13:59:36 +02:00
"../api/transport:datagram_transport_interface",
"../api/transport/media:media_transport_interface",
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 15:16:49 +02:00
"../api/transport/rtp:rtp_source",
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",
"../api/video:video_frame_i420",
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",
"../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",
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",
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",
Wire up field trials for some experimental screenshare settings Bug: b/132074409 Change-Id: I83d5334255bad4fcf585b9850506bbfe1914ba57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147868 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28758}
2019-08-02 16:32:24 +02:00
"../rtc_base/experiments:experimental_screenshare_settings",
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",
Export symbols needed by the Chromium component build (part 3). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). Bug: webrtc:9419 Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94 Reviewed-on: https://webrtc-review.googlesource.com/c/106343 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 14:13:50 +02:00
"../rtc_base/system:rtc_export",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../rtc_base/third_party/base64",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
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) {
deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
} 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",
]
}
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_data") {
Buffers non atomic message send with usrsctp lib. Currently we set the EOR bit when sending a message through the sctp library. This makes the send non atomic, meaning that message can be partially accepted by the sctp socket. Currently we ignore the sent amount result, but this change now checks that result and buffers the remaining message to be sent later in the case that it was only partially accepted by usrsctp. Bug: webrtc:10922 Change-Id: I9ff563c40e2b7dbdeb19b40d07c43a15ff7c9b49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150562 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29051}
2019-08-30 09:02:46 -07:00
defines = [
# "SCTP_DEBUG" # Uncomment for SCTP debugging.
]
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 = [
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
":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
"..:webrtc_common",
Move webrtc/audio_*.h to webrtc/api/call BUG=webrtc:5878 NOTRY=True Review-Url: https://codereview.webrtc.org/2059703002 Cr-Commit-Position: refs/heads/master@{#13996}
2016-08-31 07:33:05 -07:00
"../api:call_api",
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies. transport.h defines an interface for sending rtp and rtcp packets, which is used by MediaChannel in webrtc/media/engine, {Audio|Video}{Send|Receive}Stream and in a few other places. It was part of the build target //webrtc:webrtc, which is a monolithic target with all webrtc production code. This CL moves the header to its own target in webrtc/api and deprecates the old location. Targets in webrtc/api should in general only depend on other targets in webrtc/api. The target webrtc/api:call_api depends on transport.h. This change also makes webrtc/voice_engine pass GN's header include checker and is needed in order for webrtc/api:call_api to pass it. transport.h will be completely removed in a follow-up CL in a few weeks after clients have updated their includes. NOTRY=True BUG=webrtc:5589, webrtc:5878, webrtc:6785 Review-Url: https://codereview.webrtc.org/2426563003 Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 07:02:13 -08:00
"../api:transport_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
"../p2p:rtc_p2p",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base",
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",
"../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
"../system_wrappers",
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",
Clarify expectation on GlobalLock Merge GlobalLock and GlobalLockPod, make member private. annotate creation of all GlobalLocks with ABSL_CONST_INIT Bug: None Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29447}
2019-10-10 11:12:15 +02:00
"//third_party/abseil-cpp/absl/base:core_headers",
Buffers non atomic message send with usrsctp lib. Currently we set the EOR bit when sending a message through the sctp library. This makes the send non atomic, meaning that message can be partially accepted by the sctp socket. Currently we ignore the sent amount result, but this change now checks that result and buffers the remaining message to be sent later in the case that it was only partially accepted by usrsctp. Bug: webrtc:10922 Change-Id: I9ff563c40e2b7dbdeb19b40d07c43a15ff7c9b49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150562 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29051}
2019-08-30 09:02:46 -07:00
"//third_party/abseil-cpp/absl/types:optional",
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} The only thing that differs from the previous attempt in https://codereview.webrtc.org/1979933002/ is that none of the new targets are not hooked up to the webrtc target in webrtc/BUILD.gn, which should make it not break the chromium.webrtc.fyi bots. Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} in preparation for removing src/third_party/libjingle in Chromium. Changes between previous attempt and the one before that (https://codereview.webrtc.org/1973313002) are: * Added libstunprober target * Adjusted warnings for Chromium's clang plugins * webrtc/pc/externalhmac.{h,cc} added for Chromium builds. BUG=webrtc:4256 NOTRY=True NOPRESUBMIT=True TBR=perkj@webrtc.org, tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2037983002 Cr-Commit-Position: refs/heads/master@{#13030}
2016-06-03 03:09:32 -07:00
]
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
if (rtc_enable_sctp) {
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
"sctp/sctp_transport.cc",
"sctp/sctp_transport.h",
"sctp/sctp_transport_internal.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
]
Allow building without sctp on mac/ios. Bug: webrtc:8590 Change-Id: I4388a2292d3d5dcdb5fdcc11a1807e4790261565 Reviewed-on: https://webrtc-review.googlesource.com/27985 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21077}
2017-12-01 15:26:41 +01:00
} else {
# libtool on mac does not like empty targets.
sources = [
Fixing package boundary violation. TBR=solenberg@webrtc.org Bug: None Change-Id: I8ea08b97a4251fa9904e90c5c1f3095ea1c90a07 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/32361 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21237}
2017-12-13 07:40:54 +01:00
"sctp/noop.cc",
Allow building without sctp on mac/ios. Bug: webrtc:8590 Change-Id: I4388a2292d3d5dcdb5fdcc11a1807e4790261565 Reviewed-on: https://webrtc-review.googlesource.com/27985 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21077}
2017-12-01 15:26:41 +01:00
]
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
}
if (rtc_enable_sctp && rtc_build_usrsctp) {
include_dirs = [
# TODO(jiayl): move this into the public_configs of
# //third_party/usrsctp/BUILD.gn.
"//third_party/usrsctp/usrsctplib",
]
deps += [ "//third_party/usrsctp" ]
}
}
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.
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
deps = [
Support building WebRTC without audio and video. This CL makes the WebRTC more modular and allows the users to build WebRTC without audio and video(DataChannel only). The BUILD files in call/, logging/, media/ and pc/ are modified to support modular WebRTC. The dependencies on Call and RtcEventLog are removed from the PeerConnection. Instead of being created internally, they would be passed in by the PeerConnectionFactory. Add the CreateModularPeerConnectionFactory function which allow the users to create a PeerConnectionFactory with the modules they need. If the users want to build WebRTC without audio and video, they can pass in null pointers for modules they don't need. (MediaEngine, VideoEncoderFactory etc.) BUG=webrtc:7613 Review-Url: https://codereview.webrtc.org/2854123003 Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 12:52:32 -07:00
":rtc_audio_video",
":rtc_data",
]
}
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",
[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",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame_i420",
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",
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 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
]
}
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
rtc_media_unittests_resources = [
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 06:47:31 +02:00
"../resources/media/captured-320x240-2s-48.frames",
"../resources/media/faces.1280x720_P420.yuv",
"../resources/media/faces_I420.jpg",
"../resources/media/faces_I422.jpg",
"../resources/media/faces_I444.jpg",
"../resources/media/faces_I411.jpg",
"../resources/media/faces_I400.jpg",
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
]
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
if (is_ios) {
bundle_data("rtc_media_unittests_bundle_data") {
testonly = true
sources = rtc_media_unittests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
}
GN: Introduce templates. Defines the rtc_executable, rtc_source_set, rtc_test and rtc_static_library templates. These templates provide no functionality yet, but will enable common configuration to be introduced, avoiding repetition in every target Changes summary: - Prepend rtc_ to test, source_set, executable and static_library targets - Change "configs -= [" to "suppressed_configs += [" - Include webrtc/build/webrtc.gni where it wasn't included yet - Delete import("//testing/test.gni"), since rtc_test makes it unnecessary. BUG=webrtc:6187 TBR=henrik.lundin@webrtc.org,tommi@webrtc.org NOTRY=True Review-Url: https://codereview.webrtc.org/2301053002 Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 04:10:34 -07:00
rtc_test("rtc_media_unittests") {
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
Skip unit test if GYP_DEFINES="rtc_use_h264=1" is not set. Unit test would fail in default configuration (e.g. rtc_use_h264=0), cause it tests instantiating H264 specifics. BUG=webrtc:6194, webrtc:6198 Review-Url: https://codereview.webrtc.org/2228733004 Cr-Commit-Position: refs/heads/master@{#13929}
2016-08-26 02:59:47 -07:00
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",
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_constants",
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_data",
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_encoder_simulcast_proxy",
":rtc_internal_video_codecs",
":rtc_media",
":rtc_media_base",
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
":rtc_media_engine_defaults",
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_tests_utils",
":rtc_simulcast_encoder_adapter",
":rtc_vp9_profile",
[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
"../:webrtc_common",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:create_simulcast_test_fixture_api",
Add support for target rate constraints WebRTC video engine now configures bitrate on media transport correctly. Bug: webrtc:9719 Change-Id: I85884cd76644b7eca3763cec8ce9e31b5b64db27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127941 Commit-Queue: Peter Slatala <psla@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27167}
2019-03-18 10:25:02 -07:00
"../api:fake_media_transport",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../api:libjingle_peerconnection_api",
"../api:mock_video_bitrate_allocator",
"../api:mock_video_bitrate_allocator_factory",
"../api:mock_video_codec_factory",
Consolidate mock video encoders Use the MockVideoEncoder from the api target in encoder_simulcast_proxy_unittest, rather than a custom MockEncoder. This also prevents issue when new SetRates() is made pure virtual. Bug: webrtc:10481 Change-Id: I72469803c00f7014eeac5b9321d1e0d716fa245d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134211 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27774}
2019-04-25 16:01:03 +02:00
"../api:mock_video_encoder",
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",
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:simulcast_test_fixture_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
Delete deprecated rtc_event_log header Bug: webrtc:10206 Change-Id: I9ed3148843c647372993729b87c0e74741ab540b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28791}
2019-08-07 12:24:53 +02:00
"../api/rtc_event_log",
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",
"../api/task_queue:default_task_queue_factory",
Use function_video_(en|de)coder_factory from api Remove them from test. It is completion of the move started with https://webrtc-review.googlesource.com/c/src/+/107705 Bug: None Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6 Reviewed-on: https://webrtc-review.googlesource.com/c/107891 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 15:57:48 +02:00
"../api/test/video:function_video_factory",
Move MediaTransportInterface out of the libjingle_peerconnection_api target And move related files into api/transport/ and api/transport/media/. The moved files are unchanged, except that congestion_control_interface.h and datagram_transport_interface.h no longer include media_transport_interface.h, instead, they forward declare the few MediaTransport* types they reference. Bug: webrtc:8733 Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-12 13:59:36 +02:00
"../api/transport/media:media_transport_interface",
Add GetSources to VideoRtpReceiver BUG=webrtc:9770 Change-Id: I16143fce6eb727bbab0f6c621aa5b51bc6d28d6b Reviewed-on: https://webrtc-review.googlesource.com/101600 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24858}
2018-09-26 16:04:32 +02:00
"../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:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocation",
"../api/video:video_frame",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../api/video:video_frame_i420",
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: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",
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",
Add missing files to audio_processing. Bug: webrtc:7621 Change-Id: I2cab764232fc4e084ed8a489f4cf3a3ac562c894 Reviewed-on: https://webrtc-review.googlesource.com/34658 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21461}
2017-12-19 16:44:45 +01:00
"../modules/audio_processing:mocks",
webrtcvideoengine_unittest: Use RtpHeaderParser class for parsing rtp header. Removes parsing code in ParseRtpPacket. Bug: none Change-Id: I190b3f8abd5f1881cf19e14b7f31cc8f85b5f156 Reviewed-on: https://webrtc-review.googlesource.com/86823 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23829}
2018-07-03 10:40:40 +02:00
"../modules/rtp_rtcp",
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:simulcast_test_fixture_impl",
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",
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:webrtc_vp8",
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:p2p_test_utils",
"../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",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../rtc_base:rtc_base_tests_utils",
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",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../rtc_base/third_party/sigslot",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../test:audio_codec_mocks",
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
"../test:field_trial",
Make the RtpHeaderParserImpl available to tests and tools only. There are a few reasons for making this test only: * The code is only used by tests and utilities. * The pure interface has only a single implementation so an interface isn't really needed. (a followup change could remove it altogether) * The implementation always incorporates locking regardless of how the class gets used. See e.g. previous use in the Packet class. * The implementation is a layer on top of RtpUtility::RtpHeaderParser which is sufficient for most production cases. Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36 Bug: none Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 16:39:05 +02:00
"../test:rtp_test_utils",
Delete rtc_base/unittest_main.cc Usage replaced with test/test_main.cc. Bug: webrtc:5996 Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28010}
2019-05-21 13:01:58 +02:00
"../test:test_main",
merge deps= and unconditional deps+= blocks in media/BUILD.gn no dependencies added or removed, only resorted. Bug: None Change-Id: Iaaee4ca93d1e067c2a5552bb8e5861893f5542e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132320 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27540}
2019-04-10 12:50:24 +02:00
"../test:test_support",
"../test:video_test_common",
"//third_party/abseil-cpp/absl/algorithm:container",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Depend directly on destination targets. Makes 'gn check' happy. Followup to https://webrtc-review.googlesource.com/c/src/+/106820 Bug: webrtc:5876, webrtc:9855 Change-Id: I33fa2c31ba26dc10c9a9c17da0ffed255c1f4d5b Reviewed-on: https://webrtc-review.googlesource.com/c/108760 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25447}
2018-10-30 22:23:02 +01:00
"//third_party/abseil-cpp/absl/strings",
Delete rtc_base/unittest_main.cc Usage replaced with test/test_main.cc. Bug: webrtc:5996 Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28010}
2019-05-21 13:01:58 +02:00
"//third_party/abseil-cpp/absl/types:optional",
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 = [
"base/codec_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"base/rtp_data_engine_unittest.cc",
"base/rtp_utils_unittest.cc",
"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",
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
"engine/encoder_simulcast_proxy_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"engine/internal_decoder_factory_unittest.cc",
"engine/multiplex_codec_factory_unittest.cc",
"engine/null_webrtc_video_engine_unittest.cc",
WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs. Reland of https://codereview.webrtc.org/2072753002/ which broke chromium due to how their build was setup. This issue should now be resolved. Changed WebRtcVoiceEngine to present receive codecs from the formats provided by its decoder factory. Added supported formats to BuiltinAudioDecoderFactory. Added helper functions for creating some simple decoder factories for mocking. Created a PayloadTypeMapper for assigning payload types to formats. I think we'll eventually want to use this further up, or possibly in both the audio and video sides. It would be best if the engines didn't have to talk payload types at all, but it might be more difficult to get around when payload types depend on each-other, like the RTX codecs for video. BUG=webrtc:5805 TBR=ivoc@webrtc.org Review-Url: https://codereview.webrtc.org/2250683002 Cr-Commit-Position: refs/heads/master@{#13793}
2016-08-17 02:45:41 -07:00
"engine/payload_type_mapper_unittest.cc",
Remove webrtc::VideoEncoderFactory Replace the use of webrtc::VideoEncoderFactory with cricket::WebRtcVideoEncoderFactory and remove the adapter classes between these two factory types. Some code changes were necessary in order to accomplish this: * Move SimulcastEncoderAdapter from webrtc/modules/video_coding/codecs/vp8 to webrtc/media/engine (that's where it's used). * Rename simulcast_unittest.h to simulcast_test_utility.h and make it into it's own target, because it's used from both simulcast_unittest.cc and simulcast_encoder_adapter_unittest.cc. * Remove ownership of the encoder factory from SimulcastEncoderAdapter, and make the necessary changes in surrounding code. The goal with this CL is to clean up the code, and also to free up the name webrtc::VideoEncoderFactory for future use. BUG=webrtc:7925 Review-Url: https://codereview.webrtc.org/2964953002 Cr-Commit-Position: refs/heads/master@{#18945}
2017-07-10 03:26:36 -07:00
"engine/simulcast_encoder_adapter_unittest.cc",
Add unit tests for simulcast layer configurations. Bug: webrtc:9341 Change-Id: I82b164bd1faca2042be6176463c930b693d951eb Reviewed-on: https://webrtc-review.googlesource.com/83321 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23922}
2018-06-27 10:44:56 +02:00
"engine/simulcast_unittest.cc",
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_unittest.cc",
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 09:11:00 -08:00
"engine/webrtc_media_engine_unittest.cc",
"engine/webrtc_video_engine_unittest.cc",
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
]
Enable more unittests on iOS, and disable those that fail on simulator Tests enabled: * modules_unittests * ortc_unittests * rtc_media_unittests * rtc_unittests * video_capture_tests * video_engine_tests BUG=webrtc:5566,webrtc:4752,webrtc:5568,webrtc:5569 Review-Url: https://codereview.webrtc.org/2938193002 Cr-Commit-Position: refs/heads/master@{#18710}
2017-06-22 01:47:20 -07:00
# TODO(kthelgason): Reenable this test on iOS.
# See bugs.webrtc.org/5569
if (!is_ios) {
(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
sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
Enable more unittests on iOS, and disable those that fail on simulator Tests enabled: * modules_unittests * ortc_unittests * rtc_media_unittests * rtc_unittests * video_capture_tests * video_engine_tests BUG=webrtc:5566,webrtc:4752,webrtc:5568,webrtc:5569 Review-Url: https://codereview.webrtc.org/2938193002 Cr-Commit-Position: refs/heads/master@{#18710}
2017-06-22 01:47:20 -07:00
}
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. This allows building without SCTP support (and even building/running tests). The "HAVE_SCTP" define has been functional for a while, but there wasn't any easy way to turn it on/off. NOTRY=True BUG=webrtc:6933 Review-Url: https://codereview.webrtc.org/2593313002 Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 10:53:38 -08:00
if (rtc_enable_sctp) {
Added usrsctp reliablitiy stress test. The reliability test is intended to reproduce a deadlock issue in usrsctp known as https://github.com/sctplab/usrsctp/issues/325. Bug: None Change-Id: If3c1ff70f41dcc5ec27a4fb529eb4cc6fd343a48 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157107 Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Seth Hampson <shampson@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29617}
2019-10-25 12:48:11 +03:00
sources += [
"sctp/sctp_transport_reliability_unittest.cc",
"sctp/sctp_transport_unittest.cc",
]
Hook up new "rtc_enable_sctp" build argument to "HAVE_SCTP" define. This allows building without SCTP support (and even building/running tests). The "HAVE_SCTP" define has been functional for a while, but there wasn't any easy way to turn it on/off. NOTRY=True BUG=webrtc:6933 Review-Url: https://codereview.webrtc.org/2593313002 Cr-Commit-Position: refs/heads/master@{#15763}
2016-12-22 10:53:38 -08:00
}
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
if (rtc_opus_support_120ms_ptime) {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
} else {
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
}
GN: Declare resources for targets. Declare resources for GN targets so that they can be isolated NOTRY=True BUG=chromium:497757 Review-Url: https://codereview.webrtc.org/2340753002 Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 05:10:01 -07:00
data = rtc_media_unittests_resources
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 (is_android) {
deps += [ "//testing/android/native_test:native_test_support" ]
GN: Synchronize resources between Android and iOS. iOS tests packaged into an .app uses the same way of defining resources (the data attribute). Some iOS simulator tests are failing due to missing resources, so let's sync them all. BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2277753003 Cr-Commit-Position: refs/heads/master@{#13898}
2016-08-24 07:48:42 -07:00
shard_timeout = 900
}
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
if (is_ios) {
deps += [ ":rtc_media_unittests_bundle_data" ]
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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