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webrtc_m130/modules/video_coding/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
rtc_static_library("encoded_frame") {
visibility = [ "*" ]
sources = [
"encoded_frame.cc",
"encoded_frame.h",
]
deps = [
":video_codec_interface",
"../../:webrtc_common",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
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",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
"../../modules:module_api",
"../../modules:module_api_public",
"../../modules/video_coding:video_coding_utility",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base/experiments:alr_experiment",
Add rtt_mult_experiment to evaluate video robustness vs. latency Bug: webrtc:9670 Change-Id: Idb4ca130bfa652b2d0bddb5bee9ed8e34c97150a Reviewed-on: https://webrtc-review.googlesource.com/96060 Commit-Queue: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24457}
2018-08-27 10:08:58 -05:00
"../../rtc_base/experiments:rtt_mult_experiment",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
"../../system_wrappers:system_wrappers",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Moved NackModule and VCMPacket to their own targets Bug: webrtc:9373 Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251 Reviewed-on: https://webrtc-review.googlesource.com/81744 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-07 16:15:40 +02:00
rtc_static_library("nack_module") {
visibility = [ "*" ]
sources = [
"histogram.cc",
"histogram.h",
"nack_module.cc",
"nack_module.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":packet",
"..:module_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../system_wrappers",
"../utility:utility",
]
}
rtc_static_library("packet") {
visibility = [ "*" ]
sources = [
"packet.cc",
"packet.h",
]
deps = [
"..:module_api",
Make RtpGenericFrameDescriptor available for E2EE. This CL makes the RtpGenericFrameDescriptor available in RTPSenderVideo::SendVideo for encryption and in RtpVideoStreamReceiver::OnReceivedFrame for decryption. Bug: webrtc:9361 Change-Id: I5b6d10138c0874657862f103c8c9a2328e6d4a66 Reviewed-on: https://webrtc-review.googlesource.com/102720 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24929}
2018-10-02 13:55:47 +02:00
"../rtp_rtcp:rtp_rtcp_format",
Moved NackModule and VCMPacket to their own targets Bug: webrtc:9373 Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251 Reviewed-on: https://webrtc-review.googlesource.com/81744 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-07 16:15:40 +02:00
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("video_coding") {
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 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
deps = [
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
]
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
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"codec_timer.cc",
"codec_timer.h",
Split VCMCodecDataBase into VCMEncoderDataBase and VCMDecoderDataBase. Intended to ease further refactoring, cleanup and deletion in this code. Bug: webrtc:8830 Change-Id: Ib862b073e93b67b4f8eedbbf40ad3a8354a566a2 Reviewed-on: https://webrtc-review.googlesource.com/55562 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22120}
2018-02-20 16:09:48 +01:00
"decoder_database.cc",
"decoder_database.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"decoding_state.cc",
"decoding_state.h",
Split VCMCodecDataBase into VCMEncoderDataBase and VCMDecoderDataBase. Intended to ease further refactoring, cleanup and deletion in this code. Bug: webrtc:8830 Change-Id: Ib862b073e93b67b4f8eedbbf40ad3a8354a566a2 Reviewed-on: https://webrtc-review.googlesource.com/55562 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22120}
2018-02-20 16:09:48 +01:00
"encoder_database.cc",
"encoder_database.h",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"fec_controller_default.cc",
"fec_controller_default.h",
Rename fec_tables_xor.h -> fec_rate_table.h This is a slightly more descriptive name, since we only have one type of erasure code (XOR), and we only have one table. BUG=None Review-Url: https://codereview.webrtc.org/2625903004 Cr-Commit-Position: refs/heads/master@{#16032}
2017-01-12 06:16:24 -08:00
"fec_rate_table.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"frame_buffer.cc",
"frame_buffer.h",
Revert "Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )" Also disabled modules_unittest.TestFrameBuffer2.* in drmemory. This reverts commit b711f10d9683b9de6ee78186f77b225fc7ebfb8f. TBR=honghaiz@webrtc.org BUG= Review URL: https://codereview.webrtc.org/1991133003 . Cr-Commit-Position: refs/heads/master@{#12806}
2016-05-19 12:19:35 +02:00
"frame_buffer2.cc",
"frame_buffer2.h",
Packet buffer for the new jitter buffer. BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
2016-04-01 02:01:54 -07:00
"frame_object.cc",
"frame_object.h",
Revert "VCMGenericDecoder threading updates for all but Android." This reverts commit a4e71b9e7e59be21b98d63cf8cb676096d9c74b0. Reason for revert: Breaking internal project Original change's description: > VCMGenericDecoder threading updates for all but Android. > > * All methods now have thread checks. > * Variable access associated with thread checkers. > * Remove need for |rtc::CriticalSection lock_| > > Since the android decoder is inherently asynchronous, and > FrameBuffer2's decoder doesn't support posting tasks to it > yet (for async decode completion), we need to tackle android > separately. Once FrameBuffer2 gets changed to use a TaskQueue > or ProcessThread, we can move Android over to delivering decoded > frames on the right thread/queue and delete generic_decoder_android.*. > > Note: This is a subset of code that was previously reviewed here: > - https://codereview.webrtc.org/2764573002/ > > Bug: webrtc:7361, webrtc:8907, chromium:695438 > Change-Id: I118609dfa5c0f0180287d8c2b6d62987b7473c5c > Reviewed-on: https://webrtc-review.googlesource.com/55060 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22119} TBR=sakal@webrtc.org,tommi@webrtc.org Change-Id: I3afe4671f9d06bb4a2b17e4f14c21d79f773e067 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7361, webrtc:8907, chromium:695438 Reviewed-on: https://webrtc-review.googlesource.com/56282 Reviewed-by: Lu Liu <lliuu@webrtc.org> Commit-Queue: Lu Liu <lliuu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22143}
2018-02-21 19:38:59 +00:00
"generic_decoder.cc",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"generic_decoder.h",
"generic_encoder.cc",
"generic_encoder.h",
Decode h264 fmtp sprop-parameter-sets to binary. BUG=webrtc:5948 Review-Url: https://codereview.webrtc.org/2544493005 Cr-Commit-Position: refs/heads/master@{#15474}
2016-12-08 03:57:17 -08:00
"h264_sprop_parameter_sets.cc",
"h264_sprop_parameter_sets.h",
H264SpsPpsTracker class which keep tracks of SPS/PPS. The H264SpsPpsTracker class: - Keeps track of all received SPS/PPS. - Decides whether a packet should be inserted into the PacketBuffer or not. - Don't insert if this packet only contains SPS and/or PPS. - Don't insert if this is the first packet of and IDR and we have not received the required SPS/PPS. - Insert start codes, and in the case of the first packet of an IDR prepend the bitstream with the given SPS/PPS for this IDR. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/2466993003 Cr-Commit-Position: refs/heads/master@{#14906}
2016-11-03 04:03:01 -07:00
"h264_sps_pps_tracker.cc",
"h264_sps_pps_tracker.h",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"include/video_codec_initializer.h",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"include/video_coding.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"inter_frame_delay.cc",
"inter_frame_delay.h",
"internal_defines.h",
"jitter_buffer.cc",
"jitter_buffer.h",
"jitter_buffer_common.h",
"jitter_estimator.cc",
"jitter_estimator.h",
"media_opt_util.cc",
"media_opt_util.h",
"media_optimization.cc",
"media_optimization.h",
"nack_fec_tables.h",
Packet buffer for the new jitter buffer. BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
2016-04-01 02:01:54 -07:00
"packet_buffer.cc",
"packet_buffer.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"receiver.cc",
"receiver.h",
Logic for finding frame references moved from PacketBuffer to new class RtpFrameReferenceFinder. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/1961053002 Cr-Commit-Position: refs/heads/master@{#12725}
2016-05-13 06:01:03 -07:00
"rtp_frame_reference_finder.cc",
"rtp_frame_reference_finder.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"rtt_filter.cc",
"rtt_filter.h",
"session_info.cc",
"session_info.h",
"timestamp_map.cc",
"timestamp_map.h",
"timing.cc",
"timing.h",
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"video_codec_initializer.cc",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"video_coding_impl.cc",
"video_coding_impl.h",
"video_receiver.cc",
"video_sender.cc",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
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
deps += [
Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. BUG=webrtc:8423 Change-Id: I0f0d59461afead700c20c9a2ed9b2bc991590b4a Reviewed-on: https://webrtc-review.googlesource.com/15101 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20559}
2017-11-02 14:28:06 +01:00
":codec_globals_headers",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
":encoded_frame",
Moved NackModule and VCMPacket to their own targets Bug: webrtc:9373 Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251 Reviewed-on: https://webrtc-review.googlesource.com/81744 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-07 16:15:40 +02:00
":packet",
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
":video_codec_interface",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
":video_coding_utility",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
In generic encoder ensure capture timestamp is preserved. Because some hardware encoders may not preserve capture timestamp, use rtp timestamp as a key to capture time storage and restore corrupted capture timestamps whenever possible. Bug: webrtc:8497 Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4 Reviewed-on: https://webrtc-review.googlesource.com/23620 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20732}
2017-11-16 14:33:53 +01:00
"..:module_api_public",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"../../api:fec_controller_api",
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:encoded_frame",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
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",
"../../api/video:video_frame_i420",
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
"../../api/video_codecs:video_codecs_api",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove 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",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
Moved ALR experiment settings to new experiments folder. This replaces most of the existing dependencies on the application limited region(ALR) detector. This is to achieve a greater separation of concerns and will make further refactoring regarding the ALR Detector less invasive on other parts of the code base. Bug: webrtc:8415 Change-Id: I92912254c6d02285cce6a88f6789f0ac94794c88 Reviewed-on: https://webrtc-review.googlesource.com/37560 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21598}
2018-01-12 10:54:18 +01:00
"../../rtc_base/experiments:alr_experiment",
Add rtt_mult_experiment to evaluate video robustness vs. latency Bug: webrtc:9670 Change-Id: Idb4ca130bfa652b2d0bddb5bee9ed8e34c97150a Reviewed-on: https://webrtc-review.googlesource.com/96060 Commit-Queue: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24457}
2018-08-27 10:08:58 -05:00
"../../rtc_base/experiments:rtt_mult_experiment",
Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix Bug: chromium:805946 Change-Id: Ibb5dce9af27d0e48c9aee6b0a860b6f62b3c76a0 Reviewed-on: https://webrtc-review.googlesource.com/46145 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21889}
2018-02-05 10:33:35 +01:00
"../../rtc_base/system:fallthrough",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../../rtc_base/third_party/base64",
Move timestamp_extrapolator.h to rtc_base/time/ This moves it from an API directory (system_wrappers/include/) to a non-API directory, which is exactly what we want for utilities like this. BUG=webrtc:8445 Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f Reviewed-on: https://webrtc-review.googlesource.com/21540 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22569}
2018-03-22 15:29:03 +01:00
"../../rtc_base/time:timestamp_extrapolator",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
"../../system_wrappers",
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
"../rtp_rtcp:rtp_rtcp_format",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../utility:utility",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"//third_party/abseil-cpp/absl/types:variant",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
}
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
rtc_source_set("video_codec_interface") {
Make video_codec_interface visible. These headers contain public defines for the module, so it makes sense it's visible. Bug: webrtc:7620 Change-Id: I4c2604dcfb3beb4bfa0803a15d0e1a89c374e4cf Reviewed-on: https://webrtc-review.googlesource.com/39261 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21616}
2018-01-15 16:01:43 +01:00
visibility = [ "*" ]
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
sources = [
"include/video_codec_interface.h",
"include/video_coding_defines.h",
"include/video_error_codes.h",
Fix chromium warnings for SdpVideoFormat. Bug: webrtc:163 Change-Id: I29ad3c00116692f047456df7721ba636bbb2ca89 Reviewed-on: https://webrtc-review.googlesource.com/64723 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22618}
2018-03-27 08:31:45 +02:00
"video_coding_defines.cc",
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 = [
"..:module_api",
"../..:webrtc_common",
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",
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
"../../api/video_codecs:video_codecs_api",
"../../common_video:common_video",
]
}
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
rtc_source_set("mock_headers") {
testonly = true
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
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
"include/mock/mock_video_codec_interface.h",
]
deps = [
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
":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
":video_coding_utility",
"../..:webrtc_common",
"../../test:test_support",
]
}
rtc_source_set("codec_globals_headers") {
Set public visibility for rtp_rtcp and video_coding targets Though discouraged, those folders are listed in native-api NOTRY=True Bug: webrtc:9808 Change-Id: I9407c8d69a0d75196cfa9435f5e459264c64e046 Reviewed-on: https://webrtc-review.googlesource.com/c/103364 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24953}
2018-10-03 14:54:43 +02:00
visibility = [ "*" ]
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
sources = [
"codecs/h264/include/h264_globals.h",
"codecs/interface/common_constants.h",
"codecs/vp8/include/vp8_globals.h",
"codecs/vp9/include/vp9_globals.h",
]
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
deps = [
"../../rtc_base:checks",
]
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
}
rtc_source_set("video_coding_utility") {
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 = [ "*" ]
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
sources = [
Reland #2 of Issue 2434073003: Extract bitrate allocation ... This is yet another reland of https://codereview.webrtc.org/2434073003/ including two fixes: 1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that. 2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams. Please review only the changes after patch set 1. Original description: Extract bitrate allocation of spatial/temporal layers out of codec impl. This CL makes a number of intervowen changes: * Add BitrateAllocation struct, that contains a codec independent view of how the target bitrate is distributed over spatial and temporal layers. * Adds the BitrateAllocator interface, which takes a bitrate and frame rate and produces a BitrateAllocation. * A default (non layered) implementation is added, and SimulcastRateAllocator is extended to fully handle VP8 allocation. This includes capturing TemporalLayer instances created by the encoder. * ViEEncoder now owns both the bitrate allocator and the temporal layer factories for VP8. This allows allocation to happen fully outside of the encoder implementation. This refactoring will make it possible for ViEEncoder to signal the full picture of target bitrates to the RTCP module. BUG=webrtc:6301 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/2510583002 . Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 16:41:30 +01:00
"utility/default_video_bitrate_allocator.cc",
"utility/default_video_bitrate_allocator.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"utility/frame_dropper.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/frame_dropper.h",
Move VP9 frame rate controller to separate class. Bug: webrtc:9669 Change-Id: I6f30587778e9783182af11d2410464024918e171 Reviewed-on: https://webrtc-review.googlesource.com/96201 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24487}
2018-08-29 01:05:26 +02:00
"utility/framerate_controller.cc",
"utility/framerate_controller.h",
Add support for writing raw encoder output to .ivf files. Also refactor GenericEncoder to use these file writers, and remove use of preprocessor to enable file writing. BUG= Review URL: https://codereview.webrtc.org/1853813002 Cr-Commit-Position: refs/heads/master@{#12372}
2016-04-15 01:24:14 -07:00
"utility/ivf_file_writer.cc",
"utility/ivf_file_writer.h",
Refactor QualityScaler and MovingAverage The MovingAverage class was very specific to the QualityScaler. This commit generalizes the MovingAverage class to be useful in other situations as well, and adapts the QualityScaler to use the new MovingAverage. BUG=webrtc:6304 Review-Url: https://codereview.webrtc.org/2310853002 Cr-Commit-Position: refs/heads/master@{#14207}
2016-09-14 02:14:58 -07:00
"utility/moving_average.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/moving_average.h",
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
"utility/quality_scaler.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/quality_scaler.h",
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
"utility/simulcast_rate_allocator.cc",
"utility/simulcast_rate_allocator.h",
"utility/simulcast_utility.cc",
"utility/simulcast_utility.h",
Hardware VP8 encoding: Use QP as metric for resize. Add vp8 frame header parser to get QP from vp8 bitstream. BUG= 4273 R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49259004 Cr-Commit-Position: refs/heads/master@{#9256}
2015-05-21 11:12:02 -07:00
"utility/vp8_header_parser.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/vp8_header_parser.h",
Add vp9 QP parser. BUG=webrtc:7662 Review-Url: https://codereview.webrtc.org/2891803003 Cr-Commit-Position: refs/heads/master@{#18260}
2017-05-24 10:00:16 -07:00
"utility/vp9_uncompressed_header_parser.cc",
"utility/vp9_uncompressed_header_parser.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
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
":video_codec_interface",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. (patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ )" This reverts commit efd902cb1d9bbd81247a3e168f2080beae761d78. Originally reviewed in https://codereview.webrtc.org/2149553002 The uptream problem should be fixed by https://codereview.webrtc.org/2145393003/ BUG=webrtc:5687 TBR=tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2152013002 Cr-Commit-Position: refs/heads/master@{#13483}
2016-07-14 23:35:55 -07:00
"../../common_video",
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",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:sequenced_task_checker",
Make quality scaler downscale faster. Include dropped frames by the encoder in the frame drop percentage. To react faster at low framerates: - Use ExpFilter instead of MovingAverage to filter QP values. - Reduce sampling interval while waiting for minimum number of needed frames (when not in fast rampup mode). A separate slower ExpFilter is used for upscaling. Bug: webrtc:9169 Change-Id: If7ff6c3bd4201fda2da67125889838fe96ce7061 Reviewed-on: https://webrtc-review.googlesource.com/70761 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23014}
2018-04-24 16:53:25 +02:00
"../../rtc_base/experiments:quality_scaling_experiment",
Delete root header file typedef.h. Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 16:05:48 +02:00
"../../rtc_base/system:arch",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +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",
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
"../rtp_rtcp:rtp_rtcp_format",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_h264") {
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 = [ "*" ]
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
sources = [
"codecs/h264/h264.cc",
"codecs/h264/include/h264.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
}
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. It works on all platforms except Android and iOS (FFmpeg limitation). Implemented behind compile time flags, off by default. The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default. Flags to turn it on: - rtc_use_h264 = true - ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder) Tests using H264: - video_loopback --codec=H264 - screenshare_loopback --codec=H264 - video_engine_tests (EndToEndTest.SendsAndReceivesH264) NOTRY=True BUG=500605, 468365 BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424 Review URL: https://codereview.webrtc.org/1306813009 Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 01:36:03 -08:00
defines = []
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
deps = [
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
":video_codec_interface",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
":video_coding_utility",
Extract color space from H264 decoder Makes use of ColorSpace class to extract info from H264 stream. Bug: webrtc:9522 Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279 Reviewed-on: https://webrtc-review.googlesource.com/90180 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24085}
2018-07-23 17:55:25 -07:00
"../../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",
Move list of supported H264 codecs from InternalEncoderFactory to h264.h This CL is a clean-up to prepare for adding more supported codecs for the internal H264 SW codec. Bug: webrtc:8317 Change-Id: If483d05c01c40bbc81cbd1a6aad89961689714ef Reviewed-on: https://webrtc-review.googlesource.com/4940 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20105}
2017-09-29 17:54:09 +02:00
"../../api/video_codecs:video_codecs_api",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../../media:rtc_h264_profile_id",
Enable include check for webrtc_h264 target BUG=webrtc:7242,webrtc:6828 Review-Url: https://codereview.webrtc.org/2718143002 Cr-Commit-Position: refs/heads/master@{#16888}
2017-02-28 00:28:04 -08:00
"../../media:rtc_media_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",
Add missing files to rtc_base. Bug: webrtc:7640 Change-Id: Ia9b7f0c1c10765e7064be8d2758c1c2e68e667ed Reviewed-on: https://webrtc-review.googlesource.com/34649 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21355}
2017-12-19 11:50:21 +01:00
"../../rtc_base:rtc_base",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:metrics",
Use absl::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",
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
"//third_party/libyuv",
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
]
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. BUG=468365 NOTRY=True Review URL: https://codereview.webrtc.org/1601813005 Cr-Commit-Position: refs/heads/master@{#11333}
2016-01-21 03:34:40 -08:00
if (rtc_use_h264) {
Ability to disable the effects of |rtc_use_h264| with DisableRtcUseH264. Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name. The idea is to be able to turn off H264 from chromium with this function because... 1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled. 2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable. 3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not. The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory. BUG=chromium:500605, chromium:468365 NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1657273002 Cr-Commit-Position: refs/heads/master@{#11474}
2016-02-03 05:09:37 -08:00
defines += [ "WEBRTC_USE_H264" ]
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. It works on all platforms except Android and iOS (FFmpeg limitation). Implemented behind compile time flags, off by default. The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default. Flags to turn it on: - rtc_use_h264 = true - ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder) Tests using H264: - video_loopback --codec=H264 - screenshare_loopback --codec=H264 - video_engine_tests (EndToEndTest.SendsAndReceivesH264) NOTRY=True BUG=500605, 468365 BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424 Review URL: https://codereview.webrtc.org/1306813009 Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 01:36:03 -08:00
sources += [
Extract color space from H264 decoder Makes use of ColorSpace class to extract info from H264 stream. Bug: webrtc:9522 Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279 Reviewed-on: https://webrtc-review.googlesource.com/90180 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24085}
2018-07-23 17:55:25 -07:00
"codecs/h264/h264_color_space.cc",
"codecs/h264/h264_color_space.h",
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding. It works on all platforms except Android and iOS (FFmpeg limitation). Implemented behind compile time flags, off by default. The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default. Flags to turn it on: - rtc_use_h264 = true - ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder) Tests using H264: - video_loopback --codec=H264 - screenshare_loopback --codec=H264 - video_engine_tests (EndToEndTest.SendsAndReceivesH264) NOTRY=True BUG=500605, 468365 BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424 Review URL: https://codereview.webrtc.org/1306813009 Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 01:36:03 -08:00
"codecs/h264/h264_decoder_impl.cc",
"codecs/h264/h264_decoder_impl.h",
"codecs/h264/h264_encoder_impl.cc",
"codecs/h264/h264_encoder_impl.h",
]
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
deps += [
Fix GYP and GN references that are invalid in Chromium builds. There were a couple of GN and GYP references that were incorrect in Chromium builds: - GN references between WebRTC targets must be using relative paths, not absolute. - GYP references between WebRTC targets must be using the <(webrtc_root)v variable in order to be expanded to the correct path in a Chromium build. NOTRY=True TBR=hjon@webrtc.org, hbos@webrtc.org Review URL: https://codereview.webrtc.org/1681493002 Cr-Commit-Position: refs/heads/master@{#11521}
2016-02-07 23:04:26 -08:00
"../../common_video",
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
"//third_party/ffmpeg:ffmpeg",
"//third_party/openh264:encoder",
]
Silence deprecation warning for av_lockmgr_register It should not be used anymore - see https://webrtc-review.googlesource.com/39503 but it can be removed only after the DEPS roll. Bug: webrtc:8745 Change-Id: I1ee29948e99be6cb4a700b67d37f72a0747a9fc5 Reviewed-on: https://webrtc-review.googlesource.com/39505 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21612}
2018-01-15 09:44:18 +01:00
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
if (!build_with_mozilla) {
deps += [ "../../media:rtc_media_base" ]
}
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
}
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_i420") {
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 = [ "*" ]
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
Move i420 files to the right location There's also a presubmit check that disallows .. references in GYP files, which this solves. BUG=webrtc:5095 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1418753002 . Cr-Commit-Position: refs/heads/master@{#10371}
2015-10-22 12:23:11 +02:00
"codecs/i420/i420.cc",
"codecs/i420/include/i420.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
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
":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
":video_coding_utility",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
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",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove 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",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
"../../system_wrappers",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
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
rtc_static_library("webrtc_multiplex") {
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
sources = [
Update multiplex encoder to support having augmenting data attached to the video Multiplex encoder is now supporting attaching user-defined data to the video frame. This data will be sent with the video frame and thus is guaranteed to be synchronized. This is useful in cases where the data and video frame need to by synchronized such as sending information about 3D objects or camera tracking information with the video stream Multiplex Encoder with data is implemented in a modular way. A new VideoFrameBuffer type is created with the encoder. AugmentedVideoFrameBuffer holds the video frame and the data. MultiplexVideoEncoder encodes both the frame and data. Change-Id: I23263f70d111f6f1783c070edec70bd11ebb9868 Bug: webrtc:9632 Reviewed-on: https://webrtc-review.googlesource.com/92642 Commit-Queue: Tarek Hefny <tarekh@google.com> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24297}
2018-08-15 10:19:32 -07:00
"codecs/multiplex/augmented_video_frame_buffer.cc",
"codecs/multiplex/include/augmented_video_frame_buffer.h",
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
"codecs/multiplex/include/multiplex_decoder_adapter.h",
"codecs/multiplex/include/multiplex_encoder_adapter.h",
"codecs/multiplex/multiplex_decoder_adapter.cc",
"codecs/multiplex/multiplex_encoded_image_packer.cc",
Multiplex codec cleanups This CL performs some cleanups on multiplex files: - Adds more comments to factory about usage. - Moves image packer outside /include as it doesn't need to be public. - Other small lint issues. Bug: webrtc:9632 Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32 Reviewed-on: https://webrtc-review.googlesource.com/98700 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24615}
2018-09-06 15:35:55 -07:00
"codecs/multiplex/multiplex_encoded_image_packer.h",
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
"codecs/multiplex/multiplex_encoder_adapter.cc",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
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
":video_codec_interface",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
":video_coding_utility",
"..:module_api",
"../..:webrtc_common",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
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",
"../../api/video:video_frame_i420",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"../../api/video_codecs:video_codecs_api",
"../../common_video:common_video",
Add missing files to rtc_base. Bug: webrtc:7640 Change-Id: Ia9b7f0c1c10765e7064be8d2758c1c2e68e667ed Reviewed-on: https://webrtc-review.googlesource.com/34649 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21355}
2017-12-19 11:50:21 +01:00
"../../rtc_base:rtc_base",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"../../system_wrappers",
Reland "Wrap Alpha and YUV frame into one EncodedImage for transmission" This reverts commit d756fd06fed1b6c65dcb263cbd8f00ca23d72f3b. Original change's description: > Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission" > > This reverts commit 5670c86aeccc9bc1191725431de7998d21b73c07. > > Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy. > > Original change's description: > > Wrap Alpha and YUV frame into one EncodedImage for transmission > > > > With alpha channel, we observe the artifacts on the receiver side, and > > the reason is that when YUV channel has a key frame, it gives frame_buffer2 > > a chance to drop some previous frames. Then it is possible that some alpha > > frames got dropped, which break the alpha frame dependence chain. > > > > In this CL, we pack the YUV frame and alpha encoded frame together as one > > entity to solve the issue. > > > > Bug: webrtc:8773 > > Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7 > > Reviewed-on: https://webrtc-review.googlesource.com/38481 > > Commit-Queue: Qiang Chen <qiangchen@chromium.org> > > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21737} > > TBR=qiangchen@chromium.org,emircan@webrtc.org > > Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8773 > Reviewed-on: https://webrtc-review.googlesource.com/43260 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21739} TBR=deadbeef@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org Change-Id: I0d64b7e7a62e4f35aa012270d3826a23b3fb2337 Bug: webrtc:8773 Reviewed-on: https://webrtc-review.googlesource.com/43440 Commit-Queue: Qiang Chen <qiangchen@chromium.org> Reviewed-by: Qiang Chen <qiangchen@chromium.org> Cr-Commit-Position: refs/heads/master@{#21749}
2018-01-24 13:36:41 -08:00
"../rtp_rtcp:rtp_rtcp_format",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
]
}
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
# This target includes the internal SW codec.
Revert "Reland "Put internal video codec factories into separate target"" This reverts commit 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26. Reason for revert: Breaking internal builds 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} TBR=magjed@webrtc.org,andersc@webrtc.org Change-Id: I8d3b788cc9e43261b3ed6d3d52427b5e26bc827e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/35187 Reviewed-by: Lu Liu <lliuu@webrtc.org> Commit-Queue: Lu Liu <lliuu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21393}
2017-12-20 18:08:36 +00:00
rtc_static_library("webrtc_vp8") {
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 = [ "*" ]
Mark built-in software video codecs as poisonous. The goal is to make these injectable, and only VP8 and VP9 specific targets should depend on them. Bug: webrtc:7925 Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2 Reviewed-on: https://webrtc-review.googlesource.com/72082 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-24 16:39:05 +02:00
poisonous = [ "software_video_codecs" ]
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
sources = [
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
"codecs/vp8/default_temporal_layers.cc",
"codecs/vp8/default_temporal_layers.h",
Split TemporalLayers and TemporalLayers checker, clean up header. This CL is a step towards making the TemporalLayers landable in api/ : * It splits TemporalLayers from TemporalLayersChecker * It initially renames temporal_layer.h to vp8_temporal_layers.h and moved it into the include/ folder * It removes the dependency on VideoCodec, which was essentially only used to determine if screenshare_layers or default_temporal_layers should be used, and the number of temporal temporal layers to use. Subsequent CLs will make further cleanup before attempting a move to api Bug: webrtc:9012 Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea Reviewed-on: https://webrtc-review.googlesource.com/94156 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24920}
2018-10-01 18:47:03 +02:00
"codecs/vp8/include/temporal_layers_checker.h",
Revert "Reland "Put internal video codec factories into separate target"" This reverts commit 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26. Reason for revert: Breaking internal builds 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} TBR=magjed@webrtc.org,andersc@webrtc.org Change-Id: I8d3b788cc9e43261b3ed6d3d52427b5e26bc827e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/35187 Reviewed-by: Lu Liu <lliuu@webrtc.org> Commit-Queue: Lu Liu <lliuu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21393}
2017-12-20 18:08:36 +00:00
"codecs/vp8/include/vp8.h",
Split TemporalLayers and TemporalLayers checker, clean up header. This CL is a step towards making the TemporalLayers landable in api/ : * It splits TemporalLayers from TemporalLayersChecker * It initially renames temporal_layer.h to vp8_temporal_layers.h and moved it into the include/ folder * It removes the dependency on VideoCodec, which was essentially only used to determine if screenshare_layers or default_temporal_layers should be used, and the number of temporal temporal layers to use. Subsequent CLs will make further cleanup before attempting a move to api Bug: webrtc:9012 Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea Reviewed-on: https://webrtc-review.googlesource.com/94156 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24920}
2018-10-01 18:47:03 +02:00
"codecs/vp8/include/vp8_temporal_layers.h",
Hide libvpx vp8 encoder behind an interface and add mock for testing. Bug: webrtc:9809 Change-Id: I27baa0309511cbd849c09c5d063a64d1fb1fcebf Reviewed-on: https://webrtc-review.googlesource.com/c/103442 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25005}
2018-10-04 17:52:36 +02:00
"codecs/vp8/libvpx_interface.cc",
"codecs/vp8/libvpx_interface.h",
Split vp8_impl into webm_vp8_encoder and webm_vp8_decoder This work is in preparation for refactoring the TemporalLayers api. Bug: webrtc:9012 Change-Id: I01908ee034fb79996e687ff72d10178acf102321 Reviewed-on: https://webrtc-review.googlesource.com/61781 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22445}
2018-03-14 17:52:55 +01:00
"codecs/vp8/libvpx_vp8_decoder.cc",
"codecs/vp8/libvpx_vp8_decoder.h",
"codecs/vp8/libvpx_vp8_encoder.cc",
"codecs/vp8/libvpx_vp8_encoder.h",
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
"codecs/vp8/screenshare_layers.cc",
"codecs/vp8/screenshare_layers.h",
"codecs/vp8/temporal_layers.h",
Split TemporalLayers and TemporalLayers checker, clean up header. This CL is a step towards making the TemporalLayers landable in api/ : * It splits TemporalLayers from TemporalLayersChecker * It initially renames temporal_layer.h to vp8_temporal_layers.h and moved it into the include/ folder * It removes the dependency on VideoCodec, which was essentially only used to determine if screenshare_layers or default_temporal_layers should be used, and the number of temporal temporal layers to use. Subsequent CLs will make further cleanup before attempting a move to api Bug: webrtc:9012 Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea Reviewed-on: https://webrtc-review.googlesource.com/94156 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24920}
2018-10-01 18:47:03 +02:00
"codecs/vp8/temporal_layers_checker.cc",
"codecs/vp8/vp8_temporal_layers.cc",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Disable Chromium clang plugins for standalone build. Now that WebRTC has rolled the chromium_revision past http://crrev.com/284372 in r6784, clang has become the default compiler. Since WebRTC standalone code doesn't yet compile the Chromium Clang plugins enabled, this CL disables them for the parts of the code that doesn't yet pass compilation with them enabled. The buildbots are using Goma which is not yet switched over to Clang by default. That's why they're not red yet. BUG=163 TEST=Passing compile locally on Linux using: gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja -C out/Debug gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja -C out/Release gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default R=brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/16279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
}
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
deps = [
Split TemporalLayers and TemporalLayers checker, clean up header. This CL is a step towards making the TemporalLayers landable in api/ : * It splits TemporalLayers from TemporalLayersChecker * It initially renames temporal_layer.h to vp8_temporal_layers.h and moved it into the include/ folder * It removes the dependency on VideoCodec, which was essentially only used to determine if screenshare_layers or default_temporal_layers should be used, and the number of temporal temporal layers to use. Subsequent CLs will make further cleanup before attempting a move to api Bug: webrtc:9012 Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea Reviewed-on: https://webrtc-review.googlesource.com/94156 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24920}
2018-10-01 18:47:03 +02:00
":codec_globals_headers",
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
":video_codec_interface",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
":video_coding_utility",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
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",
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
"../../api/video_codecs:video_codecs_api",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"../../common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove 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",
vp8_impl.cc: Make it possible to base postproc deblocking level for arm on qp (e.g. turn off deblocking for low qp values). BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2970923002 Cr-Commit-Position: refs/heads/master@{#19300}
2017-08-09 23:37:05 -07:00
"../../rtc_base:rtc_numerics",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
# This target includes VP9 files that may be used for any VP9 codec, internal SW or external HW.
rtc_static_library("webrtc_vp9_helpers") {
sources = [
"codecs/vp9/svc_config.cc",
"codecs/vp9/svc_config.h",
"codecs/vp9/svc_rate_allocator.cc",
"codecs/vp9/svc_rate_allocator.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":video_codec_interface",
"../..:webrtc_common",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
Reland "Move class VideoCodec from common_types.h to its own api header file." This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd Differs from the original cl by not widening the type of VideoCodec::width and VideoCodec::height. Original change's description: > Move class VideoCodec from common_types.h to its own api header file. > > Bug: webrtc:7660 > Change-Id: I91f19bfc2565461328f30081f8383e136419aefb > Reviewed-on: https://webrtc-review.googlesource.com/79881 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23544} Bug: webrtc:7660 Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8 Reviewed-on: https://webrtc-review.googlesource.com/82140 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-08 14:03:44 +02:00
"../../api/video_codecs:video_codecs_api",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
"../../common_video",
"../../rtc_base:checks",
]
}
GN: Change rtc_source_set targets --> rtc_static_library This changes most non-test related rtc_source_set targets to be rtc_static_library instead. Targets without any .cc files are excluded. This should bring back the build behavior we used to have with GYP (i.e. same symbols exported in the libjingle_peerconnection.a file, which are used by some downstream projects). After doing an Android build with these changes: $ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf 00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory $ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf 00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*) 00000001 T webrtc::CreatePeerConnectionFactory() See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries for more details on this. NOTICE: This should be further cleaned up in the future, to reduce binary bloat and unnecessary linking time. Right now it's more important to restore the desired build output though. BUG=webrtc:6410, chromium:630755 Review-Url: https://codereview.webrtc.org/2361623004 Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 00:38:52 -07:00
rtc_static_library("webrtc_vp9") {
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 = [ "*" ]
Mark built-in software video codecs as poisonous. The goal is to make these injectable, and only VP8 and VP9 specific targets should depend on them. Bug: webrtc:7925 Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2 Reviewed-on: https://webrtc-review.googlesource.com/72082 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-24 16:39:05 +02:00
poisonous = [ "software_video_codecs" ]
Make sure vp9 actually gets excluded in gn as well. It was being excluded in this manner in gyp so it should be in gn as well. R=charujain@webrtc.org,kjellander@webrtc.org BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/2392053003 Cr-Commit-Position: refs/heads/master@{#14525}
2016-10-05 06:52:22 -07:00
if (rtc_libvpx_build_vp9) {
sources = [
"codecs/vp9/include/vp9.h",
"codecs/vp9/vp9_frame_buffer_pool.cc",
"codecs/vp9/vp9_frame_buffer_pool.h",
"codecs/vp9/vp9_impl.cc",
"codecs/vp9/vp9_impl.h",
]
} else {
sources = [
"codecs/vp9/vp9_noop.cc",
]
}
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
}
deps = [
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
":video_codec_interface",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
":video_coding_utility",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
"../..:webrtc_common",
Add color space information to webrtc::VideoFrame and extract from VP9 This CL is the first step for introducing color space information in webrtc. - Add ColorSpace class listing color profiles. - Add ColorSpace as a member of webrtc::VideoFrame. - Make use of this class by extracting info from VP9 decoder. Bug: webrtc:9522 Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad Reviewed-on: https://webrtc-review.googlesource.com/88540 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23988}
2018-07-16 10:01:49 -07:00
"../../api/video:video_frame",
Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder"" This is a reland of cb853c8f90d3410a7f0ce07915aa20db0329259d Original change's description: > Reland "Add Profile 2 configuration to VP9 Encoder and Decoder" > > This is a reland of fc9c4e88b5569f0d2cd1c64cbc27fd969ce2db17 > > Original change's description: > > Add Profile 2 configuration to VP9 Encoder and Decoder > > > > Bug: webrtc:9376 > > Change-Id: I4f627fb2b6c146a90cfcaa815da459b09dc00003 > > Reviewed-on: https://webrtc-review.googlesource.com/81980 > > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Jerome Jiang <jianj@google.com> > > Cr-Commit-Position: refs/heads/master@{#23917} > > Bug: webrtc:9376 > Change-Id: I21fc44865af4e381f99dbc5ae2baf4a53ce834ca > TBR: niklas.enbom@webrtc.org > Reviewed-on: https://webrtc-review.googlesource.com/88341 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23974} TBR: niklas.enbom@webrtc.org Bug: webrtc:9376 Change-Id: I90d7ebc2110b82901656df7f9331ae82ee010baf Reviewed-on: https://webrtc-review.googlesource.com/88582 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23977}
2018-07-13 17:10:00 -07:00
"../../api/video:video_frame_i010",
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
"../../api/video_codecs:video_codecs_api",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
"../../common_video",
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
"../../media:rtc_media_base",
"../../media:rtc_vp9_profile",
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",
Add missing files to rtc_base. Bug: webrtc:7640 Change-Id: Ia9b7f0c1c10765e7064be8d2758c1c2e68e667ed Reviewed-on: https://webrtc-review.googlesource.com/34649 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21355}
2017-12-19 11:50:21 +01:00
"../../rtc_base:rtc_base",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
"../../system_wrappers",
Configure frame references in VP9 encoder wrapper. Bug: webrtc:9585 Change-Id: I3f90d8f2b81556cfb5fa9123607ab0a9ade2bf3f Reviewed-on: https://webrtc-review.googlesource.com/93469 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24915}
2018-10-01 17:13:50 +02:00
"../../system_wrappers:field_trial",
Framerate controller for VP9 screen sharing. - Limit framerate by dropping frames before encoding. - The max framerate at screen sharing is set to 5fps. Bug: webrtc:9261 Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760 Reviewed-on: https://webrtc-review.googlesource.com/76921 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23316}
2018-05-18 17:31:19 +02:00
"../rtp_rtcp:rtp_rtcp_format",
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",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
]
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_tests) {
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
if (is_android) {
rtc_static_library("android_codec_factory_helper") {
sources = [
"codecs/test/android_codec_factory_helper.cc",
"codecs/test/android_codec_factory_helper.h",
]
deps = [
"../../api/video_codecs:video_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../sdk/android:native_api_base",
"../../sdk/android:native_api_codecs",
"../../sdk/android:native_api_jni",
"//base",
]
}
}
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
if (is_ios || is_mac) {
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
rtc_static_library("objc_codec_factory_helper") {
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
sources = [
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
"codecs/test/objc_codec_factory_helper.h",
"codecs/test/objc_codec_factory_helper.mm",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
]
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
deps = [
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",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
"../../api/video_codecs:video_codecs_api",
"../../media:rtc_audio_video",
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
"../../media:rtc_media_base",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
"../../modules:module_api",
Update videoprocessor tool to use new video factory interface This will also cause us to use the new Android HardwareVideoEncoder, instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately, the new HW encoder does not seem to work as good as the old (or the new encoder is more strict with return values or something). I don't think it adds much value to continue testing the deprecated encoder, so I filed a bug for fixing the new encoder, and in this CL I disabled the tests on Android. I want to remove as many places as possible where we use the old WebRtcVideoEncoderFactory interface, because it makes it more difficult to migrate to the new interface. Bug: webrtc:7925 Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9 Reviewed-on: https://webrtc-review.googlesource.com/27540 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21037}
2017-12-04 14:09:40 +01:00
"../../rtc_base:rtc_base_approved",
Create experimental Obj-C++ API. This can be used to wrap Objective-C components in C++ classes, so users can use the WebRTC C++ API directly together with the iOS specific components provided by our SDK. Bug: webrtc:8832 Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8 Reviewed-on: https://webrtc-review.googlesource.com/46162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21850}
2018-02-01 15:47:05 +01:00
"../../sdk:native_api",
Fix issues found by gn check. It turns out that some headers were not owned by any targets. These were: RTCVideoCodec.h RTCVideoCodecFactory.h RTCVideoCodecH264.h RTCVideoEncoderVP8.h RTCVideoDecoderVP8.h RTCVideoEncoderVP9.h RTCVideoDecoderVP9.h And some were owned by multiple targets, namely: RTCPeerConnectionFactory+Native.h RTCPeerConnectionFactory+Private.h RTCVideoFrameBuffer.h These have all been moved to their appropriate homes. This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files. Bug: webrtc:8855 Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7 Reviewed-on: https://webrtc-review.googlesource.com/49060 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22052}
2018-02-16 10:49:22 +01:00
"../../sdk:peerconnectionfactory_base_objc",
Re-enabling 'gn check': modules/video_coding:objc_codec_factory_helper. Bug: webrtc:8850 Change-Id: Ia00270b01fc143d470c5e814e4f31dfe2ce1fe78 Reviewed-on: https://webrtc-review.googlesource.com/54315 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22086}
2018-02-19 09:58:00 +01:00
"../../sdk:videocodec_objc",
Fix native api in preparation for native_api example. Add native api conversions for video frames and video renderer. This also requires some changes to sdk/BUILD to avoid cyclic dependencies. Bug: webrtc:8832 Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790 Reviewed-on: https://webrtc-review.googlesource.com/57142 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22340}
2018-03-07 10:32:03 +01:00
"../../sdk:videosource_objc",
Rename all objc targets to be suffixed for consistency BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/3001023003 Cr-Commit-Position: refs/heads/master@{#19489}
2017-08-24 05:43:45 -07:00
"../../sdk:videotoolbox_objc",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
rtc_source_set("simulcast_test_fixture_impl") {
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
testonly = true
sources = [
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
"utility/simulcast_test_fixture_impl.cc",
"utility/simulcast_test_fixture_impl.h",
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
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
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
":mock_headers",
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
":video_codec_interface",
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
":video_coding",
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
":video_coding_utility",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"../../:webrtc_common",
"../../api:simulcast_test_fixture_api",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
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",
"../../api/video:video_frame_i420",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"../../api/video_codecs:video_codecs_api",
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
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove 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",
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
"../../test:test_support",
]
}
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_source_set("video_codecs_test_framework") {
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
testonly = true
sources = [
Rename VideoCodecTest to VideoCodecUnitTest. The VideoCodecTest class is a fixture base class for the libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated to the VideoProcessor tests, which we colloquially refer to as the "codec test". This rename is thus to reduce this confusion. It should have no functional impact. Bug: webrtc:8448 Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad Reviewed-on: https://webrtc-review.googlesource.com/47160 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21867}
2018-02-01 15:45:38 +01:00
"codecs/test/video_codec_unittest.cc",
"codecs/test/video_codec_unittest.h",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"codecs/test/videoprocessor.cc",
"codecs/test/videoprocessor.h",
]
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. These suppressions are causing GN errors when Chromium targets are depending directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004) BUG=webrtc:4256 NOTRY=True Review-Url: https://codereview.webrtc.org/2408133008 Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-16 23:56:12 -07:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
deps = [
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
":video_codec_interface",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
":video_coding",
":video_coding_utility",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../..:webrtc_common",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../api:videocodec_test_fixture_api",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
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",
"../../api/video:video_frame_i420",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Refactor webrtc/modules/video_{capture,coding} for GN check This moves some GN check configurations out of .gn to individual targets. The now checked target is: "//webrtc/modules/video_capture/*", "//webrtc/modules/video_coding/*", BUG=webrtc:6828 NOTRY=True R=kjellander@webrtc.org Review-Url: https://codereview.webrtc.org/2555333004 Cr-Commit-Position: refs/heads/master@{#15488}
2016-12-08 09:56:16 -08:00
"../../common_video:common_video",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove 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",
VideoProcessor: make it runnable on a task queue. * Guard members with a SequencedTaskChecker. * Intercept encoder/decoder callbacks, and post onto task queue if needed. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2996253002 Cr-Commit-Position: refs/heads/master@{#19428}
2017-08-21 06:44:16 -07:00
"../../rtc_base:rtc_task_queue",
Create/destroying codecs on task queue + switch to TaskQueueForTest. After https://webrtc-review.googlesource.com/c/src/+/70740, we are creating/destroying the codecs on a task queue in the VideoStreamEncoder. This CL updates the VideoCodecTest to do the same. Also, this CL switches from manually Wait()'ing on the task queue to using TaskQueueForTest::SendTask. Bug: None Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e Reviewed-on: https://webrtc-review.googlesource.com/76800 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 09:38:42 +02:00
"../../rtc_base:rtc_task_queue_for_test",
VideoProcessor: make it runnable on a task queue. * Guard members with a SequencedTaskChecker. * Intercept encoder/decoder callbacks, and post onto task queue if needed. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2996253002 Cr-Commit-Position: refs/heads/master@{#19428}
2017-08-21 06:44:16 -07:00
"../../rtc_base:sequenced_task_checker",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
"../../test:test_support",
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
"../../test:video_test_common",
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002 BUG=webrtc:7037 NOTRY=True Review-Url: https://codereview.webrtc.org/2666113003 Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 02:21:11 -08:00
"../../test:video_test_support",
Reland "Updated analysis in videoprocessor." This is a reland of 1880c7162bd3637c433f9421c798808cd6eacaf7 Original change's description: > Updated analysis in videoprocessor. > > - Run analysis after all frames are processed. Before part of it was > done at bitrate change points; > - Analysis is done for whole stream as well as for each rate update > interval; > - Changed units from number of frames to time units for some metrics > and thresholds. E.g. 'num frames to hit tagret bitrate' is changed to > 'time to reach target bitrate, sec'; > - Changed data type of FrameStatistic::max_nalu_length (renamed to > max_nalu_size_bytes) from rtc::Optional to size_t. There it no need to > use such advanced data type in such low level data structure. > > Bug: webrtc:8524 > Change-Id: Ic9f6eab5b15ee12a80324b1f9c101de1bf3c702f > Reviewed-on: https://webrtc-review.googlesource.com/31901 > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21653} TBR=brandtr@webrtc.org, stefan@webrtc.org Bug: webrtc:8524 Change-Id: Ie0ad7790689422ffa61da294967fc492a13b75ae Reviewed-on: https://webrtc-review.googlesource.com/40202 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21668}
2018-01-17 15:11:44 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Adding simulcast/spatial layering support to VideoProcessor. Encoded frames are preserved and decoded after all layers are encoded. Each spatial layer is decoded with separate decoder. For quality evaluation of lowres layers original input frame is downscaled with bilinear interpolation. Encoded and decoded frames are dumped into separate files. For async codecs encoded frames are passed to decoder in encode callback, as before. Bug: webrtc:8524 Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1 Reviewed-on: https://webrtc-review.googlesource.com/43381 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21844}
2018-02-01 13:25:17 +01:00
"//third_party/libyuv",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
Use frame generator in video codec unit tests. There is no need to use real video as input for encoder in unit tests. Using generator simplifies testing on mobile devices (no need to upload files to device). Bug: none Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398 Reviewed-on: https://webrtc-review.googlesource.com/64526 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:04:35 +02:00
video_coding_modules_tests_resources = []
Add MediaCodec VP tests for uncommon resolutions. Bug: None Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd Reviewed-on: https://webrtc-review.googlesource.com/72342 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23020}
2018-04-25 12:45:16 +02:00
if (is_android) {
video_coding_modules_tests_resources += [
"../../resources/foreman_128x96.yuv",
"../../resources/foreman_160x120.yuv",
"../../resources/foreman_176x144.yuv",
"../../resources/foreman_240x136.yuv",
"../../resources/foreman_320x240.yuv",
"../../resources/foreman_480x272.yuv",
]
}
Enable more VideoCodecTests on iOS. Bug: webrtc:4755 Change-Id: I403834dbe04cc3899847eb10e5595a24e6001507 Reviewed-on: https://webrtc-review.googlesource.com/76602 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23273}
2018-05-16 16:46:58 +02:00
if (!is_android) {
Cleaning up modules_tests resources. * Removed video files which were not used by any tests. * Removed ConferenceMotion_1280_720_50.yuv for mobile builds. Bug: webrtc:8936 Change-Id: I0539e9fce20470fcc2f0af84bd297faffc4b587a Reviewed-on: https://webrtc-review.googlesource.com/60942 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22359}
2018-03-09 11:04:15 +01:00
video_coding_modules_tests_resources += [
"../../resources/FourPeople_1280x720_30.yuv",
"../../resources/ConferenceMotion_1280_720_50.yuv",
]
Remove FourPeople_1280x720_30.yuv from resources for mobile builds. Uploading of the file to device causes timeouts. I removed it from resources for iOS and Android builds and disabled the test that used the file since we don't really need to run it on test bots. Bug: webrtc:8936 Change-Id: Ia5e04c4630544eca8e56826c9e89c9c9f4dcb600 Reviewed-on: https://webrtc-review.googlesource.com/58090 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22222}
2018-02-27 15:18:18 +01:00
}
Use frame generator in video codec unit tests. There is no need to use real video as input for encoder in unit tests. Using generator simplifies testing on mobile devices (no need to upload files to device). Bug: none Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398 Reviewed-on: https://webrtc-review.googlesource.com/64526 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:04:35 +02:00
num_video_coding_modules_tests_resources = 0
foreach(i, video_coding_modules_tests_resources) {
num_video_coding_modules_tests_resources += 1
}
if (num_video_coding_modules_tests_resources > 0) {
if (is_ios || is_mac) {
bundle_data("video_coding_modules_tests_resources_bundle_data") {
testonly = true
sources = video_coding_modules_tests_resources
outputs = [
"{{bundle_resources_dir}}/{{source_file_part}}",
]
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
}
}
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
rtc_source_set("videocodec_test_impl") {
testonly = true
sources = [
"codecs/test/videocodec_test_fixture_impl.cc",
"codecs/test/videocodec_test_fixture_impl.h",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"codecs/test/videocodec_test_stats_impl.cc",
"codecs/test/videocodec_test_stats_impl.h",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
]
deps = [
":video_codec_interface",
":video_codecs_test_framework",
":video_coding_utility",
":webrtc_vp9_helpers",
"../..:webrtc_common",
"../../api:videocodec_test_fixture_api",
"../../api/video_codecs:video_codecs_api",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../call:video_stream_api",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../common_video",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../media:rtc_audio_video",
"../../media:rtc_h264_profile_id",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../media:rtc_internal_video_codecs",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../media:rtc_media_base",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../rtc_base:checks",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../rtc_base:rtc_base_approved",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../rtc_base:rtc_base_tests_utils",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../rtc_base:rtc_event",
Create/destroying codecs on task queue + switch to TaskQueueForTest. After https://webrtc-review.googlesource.com/c/src/+/70740, we are creating/destroying the codecs on a task queue in the VideoStreamEncoder. This CL updates the VideoCodecTest to do the same. Also, this CL switches from manually Wait()'ing on the task queue to using TaskQueueForTest::SendTask. Bug: None Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e Reviewed-on: https://webrtc-review.googlesource.com/76800 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 09:38:42 +02:00
"../../rtc_base:rtc_task_queue_for_test",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../system_wrappers",
"../../test:fileutils",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../test:test_common",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../test:test_support",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../test:video_test_common",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../test:video_test_support",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../rtp_rtcp:rtp_rtcp_format",
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",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
rtc_source_set("video_coding_modules_tests") {
testonly = true
sources = [
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
"codecs/h264/test/h264_impl_unittest.cc",
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
"codecs/multiplex/test/multiplex_adapter_unittest.cc",
Codec instantiation tests for testing device capabilities in batch. Change-Id: I465fb165bec9d6501015ec57d13fb556c4cb532b Bug: None Reviewed-on: https://webrtc-review.googlesource.com/58643 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22284}
2018-03-01 11:39:36 +01:00
"codecs/test/video_encoder_decoder_instantiation_tests.cc",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"codecs/test/videocodec_test_libvpx.cc",
"codecs/test/videocodec_test_parameterized.cc",
Hide libvpx vp8 encoder behind an interface and add mock for testing. Bug: webrtc:9809 Change-Id: I27baa0309511cbd849c09c5d063a64d1fb1fcebf Reviewed-on: https://webrtc-review.googlesource.com/c/103442 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25005}
2018-10-04 17:52:36 +02:00
"codecs/vp8/test/mock_libvpx_interface.h",
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
"codecs/vp8/test/vp8_impl_unittest.cc",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
"codecs/vp9/test/vp9_impl_unittest.cc",
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
]
Move H.264 codec test activation directly behind build flag. Bug: None Change-Id: I5dd366aba16b440a8e99826e116db0d6b68bf844 Reviewed-on: https://webrtc-review.googlesource.com/76621 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23231}
2018-05-14 17:11:49 +02:00
if (rtc_use_h264) {
sources += [ "codecs/test/videocodec_test_openh264.cc" ]
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
deps = [
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
":video_codecs_test_framework",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
":video_coding_utility",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
":videocodec_test_impl",
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ ) Reason for revert: The issue is now hopefully fixed. Original issue's description: > Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ ) > > Reason for revert: > Let's revert this while we investigate a problem in H264 bitstream parser. > > Original issue's description: > > Add QP for FFmpeg H264 decoder. > > > > BUG=webrtc:6541 > > > > Review-Url: https://codereview.webrtc.org/2649133007 > > Cr-Commit-Position: refs/heads/master@{#16942} > > Committed: https://chromium.googlesource.com/external/webrtc/+/879f4f6c3148fda61e988cd085e0520dc393a2af > > TBR=sprang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:6541, chromium:697795 > > Review-Url: https://codereview.webrtc.org/2726973003 > Cr-Commit-Position: refs/heads/master@{#16974} > Committed: https://chromium.googlesource.com/external/webrtc/+/4c6df8893eb4aeea1fddb2b369bb9e4743a933f6 TBR=sprang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:6541, chromium:697795 Review-Url: https://codereview.webrtc.org/2735733002 Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 03:52:55 -08:00
":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
":webrtc_multiplex",
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
":webrtc_vp8",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
":webrtc_vp9",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../..:webrtc_common",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../api:create_videocodec_test_fixture_api",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
"../../api:mock_video_codec_factory",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../api:videocodec_test_fixture_api",
Add color space information to webrtc::VideoFrame and extract from VP9 This CL is the first step for introducing color space information in webrtc. - Add ColorSpace class listing color profiles. - Add ColorSpace as a member of webrtc::VideoFrame. - Make use of this class by extracting info from VP9 decoder. Bug: webrtc:9522 Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad Reviewed-on: https://webrtc-review.googlesource.com/88540 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23988}
2018-07-16 10:01:49 -07:00
"../../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",
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",
Stop using public_deps in api/. Bug: webrtc:8603 Change-Id: I12f91365fa9a3ed12be1a9658999b6f6aad7e5e3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33220 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21584}
2018-01-11 10:13:56 +01:00
"../../api/video_codecs:video_codecs_api",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../../common_video",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../../media:rtc_h264_profile_id",
Fix build errors when rtc_use_builtin_sw_codecs is set to false. The previous effort of building WebRTC without SW codecs stopped when libjingle_peerconnection was possible to build. In order to make the group("default") target build, this basically updates a bunch of tests to explicitly depend on the built-in software video codecs. Bug: webrtc:7925 Change-Id: I2715414770c197fca01cb8dbde173a21f4434500 Reviewed-on: https://webrtc-review.googlesource.com/70503 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23216}
2018-05-14 10:11:42 +02:00
"../../media:rtc_internal_video_codecs",
Add support for H.264 constrained high profile in VideoProcessor. BUG=webrtc:8448 Change-Id: I968d6cd78dd4f3c19a7944ae4cc73c5eddb9a949 Reviewed-on: https://webrtc-review.googlesource.com/16160 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20466}
2017-10-27 14:51:46 +02:00
"../../media:rtc_media_base",
Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder"" This is a reland of cb853c8f90d3410a7f0ce07915aa20db0329259d Original change's description: > Reland "Add Profile 2 configuration to VP9 Encoder and Decoder" > > This is a reland of fc9c4e88b5569f0d2cd1c64cbc27fd969ce2db17 > > Original change's description: > > Add Profile 2 configuration to VP9 Encoder and Decoder > > > > Bug: webrtc:9376 > > Change-Id: I4f627fb2b6c146a90cfcaa815da459b09dc00003 > > Reviewed-on: https://webrtc-review.googlesource.com/81980 > > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Jerome Jiang <jianj@google.com> > > Cr-Commit-Position: refs/heads/master@{#23917} > > Bug: webrtc:9376 > Change-Id: I21fc44865af4e381f99dbc5ae2baf4a53ce834ca > TBR: niklas.enbom@webrtc.org > Reviewed-on: https://webrtc-review.googlesource.com/88341 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23974} TBR: niklas.enbom@webrtc.org Bug: webrtc:9376 Change-Id: I90d7ebc2110b82901656df7f9331ae82ee010baf Reviewed-on: https://webrtc-review.googlesource.com/88582 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23977}
2018-07-13 17:10:00 -07:00
"../../media:rtc_vp9_profile",
Add missing files to rtc_base. Bug: webrtc:7640 Change-Id: Ia9b7f0c1c10765e7064be8d2758c1c2e68e667ed Reviewed-on: https://webrtc-review.googlesource.com/34649 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21355}
2017-12-19 11:50:21 +01:00
"../../rtc_base:rtc_base",
Configure frame references in VP9 encoder wrapper. Bug: webrtc:9585 Change-Id: I3f90d8f2b81556cfb5fa9123607ab0a9ade2bf3f Reviewed-on: https://webrtc-review.googlesource.com/93469 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24915}
2018-10-01 17:13:50 +02:00
"../../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../test:test_common",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
"../../test:test_support",
"../../test:video_test_common",
Framerate controller for VP9 screen sharing. - Limit framerate by dropping frames before encoding. - The max framerate at screen sharing is set to 5fps. Bug: webrtc:9261 Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760 Reviewed-on: https://webrtc-review.googlesource.com/76921 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23316}
2018-05-18 17:31:19 +02:00
"../rtp_rtcp:rtp_rtcp_format",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': 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: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder"" This is a reland of cb853c8f90d3410a7f0ce07915aa20db0329259d Original change's description: > Reland "Add Profile 2 configuration to VP9 Encoder and Decoder" > > This is a reland of fc9c4e88b5569f0d2cd1c64cbc27fd969ce2db17 > > Original change's description: > > Add Profile 2 configuration to VP9 Encoder and Decoder > > > > Bug: webrtc:9376 > > Change-Id: I4f627fb2b6c146a90cfcaa815da459b09dc00003 > > Reviewed-on: https://webrtc-review.googlesource.com/81980 > > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Jerome Jiang <jianj@google.com> > > Cr-Commit-Position: refs/heads/master@{#23917} > > Bug: webrtc:9376 > Change-Id: I21fc44865af4e381f99dbc5ae2baf4a53ce834ca > TBR: niklas.enbom@webrtc.org > Reviewed-on: https://webrtc-review.googlesource.com/88341 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23974} TBR: niklas.enbom@webrtc.org Bug: webrtc:9376 Change-Id: I90d7ebc2110b82901656df7f9331ae82ee010baf Reviewed-on: https://webrtc-review.googlesource.com/88582 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23977}
2018-07-13 17:10:00 -07:00
"//third_party/libyuv",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
]
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
data = video_coding_modules_tests_resources
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
if (is_android) {
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
sources += [ "codecs/test/videocodec_test_mediacodec.cc" ]
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e. Reason for revert: On Lollipop Nexus 4, the 240p tests fail too. Original change's description: > Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). > > * Add audio_ prefix to CallTest::{en,de}coder_factory_. > * Let VideoQualityTest only instantiate encoders using encoder factories. > * Add HW encoder factories to VideoQualityTest. > * Add full stack tests: > - sqcif7 at 30 kbps: libvpx. > - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx. > > BUG=webrtc:8219 > TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74 > Reviewed-on: https://webrtc-review.googlesource.com/4740 > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20041} TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8219 Reviewed-on: https://webrtc-review.googlesource.com/4900 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:12 +00:00
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
deps += [ ":android_codec_factory_helper" ]
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
}
if (is_ios || is_mac) {
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
sources += [ "codecs/test/videocodec_test_videotoolbox.cc" ]
Add VideoProcessor tests verifying that H.264 keyframes contain SPS/PPS/IDR. This CL adds an EncodedFrameChecker interface which can be used by users of the VideoProcessor to inject customized per-frame checks to the encoding/decoding pipeline. This currently has two uses: - Verifying that the QP parser works correctly for VP8 and VP9, by comparing the parsed QP to that produced by libvpx. - Verifying that our H.264 encoders always produce SPS/PPS/IDR in tandem. TESTED=Galaxy S8, Pixel 2 XL, iPhone 7. BUG=webrtc:8423 Change-Id: Ic3e401546e239a9ffaf2ed2907689cebb1127805 Reviewed-on: https://webrtc-review.googlesource.com/14559 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20409}
2017-10-24 10:16:33 +02:00
Use frame generator in video codec unit tests. There is no need to use real video as input for encoder in unit tests. Using generator simplifies testing on mobile devices (no need to upload files to device). Bug: none Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398 Reviewed-on: https://webrtc-review.googlesource.com/64526 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:04:35 +02:00
deps += [ ":objc_codec_factory_helper" ]
if (num_video_coding_modules_tests_resources > 0) {
deps += [ ":video_coding_modules_tests_resources_bundle_data" ]
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
}
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
rtc_source_set("video_coding_unittests") {
testonly = true
GN: Tighten up test target visibility + refactorings Make all rtc_source_test target that contains tests that are included in a test executable only be visible to the rtc_test target. Doing this exposed a couple of errors and dependency problems that were resolved. Having this could have prevented duplicated execution of tests like the case that was recently fixed by deadbeef@ in https://codereview.webrtc.org/2820263004 New targets: * //webrtc/modules/rtp_rtcp:fec_test_helper * //webrtc/modules/rtp_rtcp:mock_rtp_rtcp * //webrtc/modules/remote_bitrate_estimator:mock_remote_bitrate_observer The mock files and targets should probably be moved into webrtc/test in the future, but that's out of the scope of this CL. BUG=webrtc:5716 NOTRY=True Review-Url: https://codereview.webrtc.org/2828793003 Cr-Commit-Position: refs/heads/master@{#17863}
2017-04-25 04:04:50 -07:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
sources = [
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"codecs/test/videocodec_test_fixture_config_unittest.cc",
"codecs/test/videocodec_test_stats_impl_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/test/videoprocessor_unittest.cc",
"codecs/vp8/default_temporal_layers_unittest.cc",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"codecs/vp8/libvpx_vp8_simulcast_test.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"codecs/vp8/screenshare_layers_unittest.cc",
Fix calculation of target bitrate of VP9 spatial layer. This fixes misprint in the code which calculates target bitrate of a VP9 spatial layer where "-" was used instead of "+". Bug: none Change-Id: I17d76a84d00e453c055c068968d7b276e9c23f51 Reviewed-on: https://webrtc-review.googlesource.com/71663 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22974}
2018-04-23 12:30:35 +02:00
"codecs/vp9/svc_config_unittest.cc",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
"codecs/vp9/svc_rate_allocator_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"decoding_state_unittest.cc",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"fec_controller_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"frame_buffer2_unittest.cc",
Implement timing frames. Timing information is gathered in EncodedImage, starting at encoders. Then it's sent using RTP header extension. In the end, it's gathered at the GenericDecoder. Actual reporting and tests will be in the next CLs. BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/2911193002 Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 07:18:55 -07:00
"generic_encoder_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"h264_sprop_parameter_sets_unittest.cc",
"h264_sps_pps_tracker_unittest.cc",
"histogram_unittest.cc",
"include/mock/mock_vcm_callbacks.h",
"jitter_buffer_unittest.cc",
"jitter_estimator_tests.cc",
"nack_module_unittest.cc",
"receiver_unittest.cc",
"rtp_frame_reference_finder_unittest.cc",
"session_info_unittest.cc",
"test/stream_generator.cc",
"test/stream_generator.h",
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
"test/test_util.h",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"timing_unittest.cc",
"utility/default_video_bitrate_allocator_unittest.cc",
"utility/frame_dropper_unittest.cc",
Move VP9 frame rate controller to separate class. Bug: webrtc:9669 Change-Id: I6f30587778e9783182af11d2410464024918e171 Reviewed-on: https://webrtc-review.googlesource.com/96201 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24487}
2018-08-29 01:05:26 +02:00
"utility/framerate_controller_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"utility/ivf_file_writer_unittest.cc",
"utility/moving_average_unittest.cc",
"utility/quality_scaler_unittest.cc",
"utility/simulcast_rate_allocator_unittest.cc",
Reland of Add experimental simulcast screen content mode The original CL was reverted because of a bug discovered by the chromium bots. Description of that CL: > Review-Url: https://codereview.webrtc.org/2636443002 > Cr-Commit-Position: refs/heads/master@{#16135} > Committed: https://chromium.googlesource.com/external/webrtc/+/a28e971e3bc8a06f366f0ef82cc7168b05a83b59 The first patch set of this CL is the same as r16135. Subsequence patch sets are the fixes applied. Some new test cases have been added, which reveal a few more bugs that have also been fixed. BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2641133002 Cr-Commit-Position: refs/heads/master@{#16299}
2017-01-26 06:12:26 -08:00
"video_codec_initializer_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"video_packet_buffer_unittest.cc",
"video_receiver_unittest.cc",
"video_sender_unittest.cc",
]
if (rtc_use_h264) {
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
sources += [
"codecs/h264/h264_encoder_impl_unittest.cc",
"codecs/h264/h264_simulcast_unittest.cc",
]
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
}
Fix build errors when rtc_use_builtin_sw_codecs is set to false. The previous effort of building WebRTC without SW codecs stopped when libjingle_peerconnection was possible to build. In order to make the group("default") target build, this basically updates a bunch of tests to explicitly depend on the built-in software video codecs. Bug: webrtc:7925 Change-Id: I2715414770c197fca01cb8dbde173a21f4434500 Reviewed-on: https://webrtc-review.googlesource.com/70503 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23216}
2018-05-14 10:11:42 +02:00
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
deps = [
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":codec_globals_headers",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
":encoded_frame",
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
":mock_headers",
Moved NackModule and VCMPacket to their own targets Bug: webrtc:9373 Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251 Reviewed-on: https://webrtc-review.googlesource.com/81744 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-07 16:15:40 +02:00
":nack_module",
":packet",
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
":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
":video_codec_interface",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":video_codecs_test_framework",
":video_coding",
":video_coding_utility",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
":videocodec_test_impl",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":webrtc_h264",
":webrtc_vp8",
":webrtc_vp9",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../..:webrtc_common",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"../../api:create_simulcast_test_fixture_api",
"../../api:simulcast_test_fixture_api",
Move VideoCodecTest configuration classes to api/test. These files are required when implementing tests based on the test fixture, and should be exposed as part of the test api. This CL also removes a usage of stringstream and fixes some chromium-style lint issues. Bug: webrtc:8982, webrtc:163 Change-Id: I132aea0da79a79587887f21897236fc9802b7574 Reviewed-on: https://webrtc-review.googlesource.com/74586 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23346}
2018-05-22 13:34:14 +02:00
"../../api:videocodec_test_fixture_api",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
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",
"../../api/video:video_frame_i420",
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../common_video:common_video",
Make Videoprocessor integration test stringly typed. This allows use of arbitrarily-named codecs. Bug: None Change-Id: If7ecbfe3ae8f08f8ebfb224219ef9192a4a0b884 Reviewed-on: https://webrtc-review.googlesource.com/69681 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22860}
2018-04-13 14:25:22 +02:00
"../../media:rtc_media_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",
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",
"../../rtc_base:rtc_base_approved",
Reland In GenericEncoder enable timing frames for encoders with internal source The original cl broke some downstream project because some internal source encoders do not call OnBitrateChanged on GenericEncoder. Bug: webrtc:9058 Change-Id: I7841c65059fb4fc9e1ab9754bb1d232ce660a990 Reviewed-on: https://webrtc-review.googlesource.com/66342 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22733}
2018-04-03 10:01:52 +02:00
"../../rtc_base:rtc_base_tests_utils",
Move sequence_number_utils.h to rtc_base/ Bug: webrtc:8440 Change-Id: I36e70da6ce70b95db7d3fce8b0013bff5c795bfc Reviewed-on: https://webrtc-review.googlesource.com/14860 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20429}
2017-10-25 13:07:09 +02:00
"../../rtc_base:rtc_numerics",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_task_queue",
Create/destroying codecs on task queue + switch to TaskQueueForTest. After https://webrtc-review.googlesource.com/c/src/+/70740, we are creating/destroying the codecs on a task queue in the VideoStreamEncoder. This CL updates the VideoCodecTest to do the same. Also, this CL switches from manually Wait()'ing on the task queue to using TaskQueueForTest::SendTask. Bug: None Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e Reviewed-on: https://webrtc-review.googlesource.com/76800 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 09:38:42 +02:00
"../../rtc_base:rtc_task_queue_for_test",
Stop using public_deps in system_wrappers. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I5e515f0e4dc955a01460d69ba4e21bdfdf152d20 Reviewed-on: https://webrtc-review.googlesource.com/29104 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21112}
2017-12-06 09:17:14 +01:00
"../../system_wrappers",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:field_trial",
Remove the public_deps to fileutils from test_support. Bug: webrtc:8946 Change-Id: Ia01d8bb1b42485e29f26792b9266228743d7fd90 No-Presubmit: true Reviewed-on: https://webrtc-review.googlesource.com/62100 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22465}
2018-03-15 15:05:39 +01:00
"../../test:fileutils",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"../../test:test_common",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"../../test:test_support",
"../../test:video_test_common",
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common. Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002 BUG=webrtc:7037 NOTRY=True Review-Url: https://codereview.webrtc.org/2666113003 Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 02:21:11 -08:00
"../../test:video_test_support",
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
"../rtp_rtcp:rtp_rtcp_format",
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",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
]
if (rtc_build_libvpx) {
deps += [ rtc_libvpx_dir ]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
}
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