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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 "Add av1 test running real video clips." This reverts commit 6958d2c6f0ce5267bdc4120d88680a4be9ed5e59. Disable the test on iOS. Bug: None Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Jerome Jiang <jianj@google.com> Cr-Commit-Position: refs/heads/master@{#31437}
2020-06-03 20:44:50 -07:00
import("//third_party/libaom/options.gni")
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
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
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("encoded_frame") {
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
visibility = [ "*" ]
sources = [
"encoded_frame.cc",
"encoded_frame.h",
]
deps = [
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
":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
":video_codec_interface",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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_public",
Cleanup includes in modules/include/module_common_types.h Add missing includes to files that were transactivly depending on removed includes. Bug: None Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 12:56:24 +02:00
"../../modules/rtp_rtcp:rtp_video_header",
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_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",
Add RTC_EXPORT to VCMEncodedFrame This is needed to be able to use webrtc::video_coding::EncodedFrame is unit tests in Chromium. TBR=tommi@webrtc.org Bug: webrtc:11380 Change-Id: Idb3b0ab667a548f5a968e02a8efd91f02585c3f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169451 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30651}
2020-02-28 16:28:34 +01:00
"../../rtc_base/system:rtc_export",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../system_wrappers",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"//third_party/abseil-cpp/absl/types:variant",
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
]
}
Introduce ChainDiffCalculator to convert flags which chains a video frame part of into chain_diffs Bug: webrtc:10342 Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31306}
2020-05-18 14:14:11 +02:00
rtc_library("chain_diff_calculator") {
sources = [
"chain_diff_calculator.cc",
"chain_diff_calculator.h",
]
deps = [
"../../rtc_base:checks",
"../../rtc_base:logging",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Introduce ChainDiffCalculator to convert flags which chains a video frame part of into chain_diffs Bug: webrtc:10342 Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31306}
2020-05-18 14:14:11 +02:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional",
]
}
Add helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
rtc_library("frame_dependencies_calculator") {
sources = [
"frame_dependencies_calculator.cc",
"frame_dependencies_calculator.h",
]
deps = [
"../../api:array_view",
"../../common_video/generic_frame_descriptor",
"../../rtc_base:checks",
"../../rtc_base:logging",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Add helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional",
]
}
NackModule2: Rename to NackRequester. The alternative new name proposed, NackTracker, is already in use in audio_coding. Fixed: webrtc:11594 Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-22 15:09:39 +02:00
rtc_library("nack_requester") {
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
sources = [
"histogram.cc",
"histogram.h",
NackModule2: Rename to NackRequester. The alternative new name proposed, NackTracker, is already in use in audio_coding. Fixed: webrtc:11594 Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-22 15:09:39 +02:00
"nack_requester.cc",
"nack_requester.h",
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
]
deps = [
"..:module_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../../api:sequence_checker",
NackModule2: coalesce repeating tasks. NackModule2 creates repeating tasks, but as there are many modules (one per receiver) these tasks execute out of phase with each other, multipliying the amount of wakeups caused. Fix this by creating a single wakeup source that serves all NackModule2 instances in a call. Bug: webrtc:12989 Change-Id: Ia9c84307eb57349679e42b673474feb2cb43f08e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226464 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34527}
2021-07-20 13:32:02 +02:00
"../../api/task_queue",
Add exponential backoff of retransmissions for a given packet Bug: webrtc:8624 Change-Id: I8900c54935bf1da11ac74665426b0d198bd6d814 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/30900 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29625}
2019-10-25 09:24:45 +02:00
"../../api/units:time_delta",
"../../api/units:timestamp",
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_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
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_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
Remove locks and dependency on ProcessThread+Module from NackModule2. Change-Id: I39975e7812d7722fd231ac57e261fd6add9de000 Bug: webrtc:11594 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175341 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31367}
2020-05-27 12:55:38 +02:00
"../../rtc_base:rtc_task_queue",
Add exponential backoff of retransmissions for a given packet Bug: webrtc:8624 Change-Id: I8900c54935bf1da11ac74665426b0d198bd6d814 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/30900 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29625}
2019-10-25 09:24:45 +02:00
"../../rtc_base/experiments:field_trial_parser",
Remove locks and dependency on ProcessThread+Module from NackModule2. Change-Id: I39975e7812d7722fd231ac57e261fd6add9de000 Bug: webrtc:11594 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175341 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31367}
2020-05-27 12:55:38 +02:00
"../../rtc_base/task_utils:pending_task_safety_flag",
"../../rtc_base/task_utils:repeating_task",
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
"../../system_wrappers",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../utility",
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
]
}
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
rtc_library("packet_buffer") {
sources = [
"packet_buffer.cc",
"packet_buffer.h",
]
deps = [
":codec_globals_headers",
"../../api:array_view",
"../../api:rtp_packet_info",
"../../api/units:timestamp",
"../../api/video:encoded_image",
"../../api/video:video_frame_type",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../rtp_rtcp:rtp_rtcp_format",
"../rtp_rtcp:rtp_video_header",
]
absl_deps = [
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/types:variant",
]
}
rtc_library("h264_packet_buffer") {
sources = [
"h264_packet_buffer.cc",
"h264_packet_buffer.h",
]
deps = [
":codec_globals_headers",
":packet_buffer",
"../../api:array_view",
"../../api:rtp_packet_info",
"../../api/units:timestamp",
"../../api/video:encoded_image",
"../../api/video:video_frame_type",
"../../common_video",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../rtp_rtcp:rtp_rtcp_format",
"../rtp_rtcp:rtp_video_header",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
Add frame_helpers A number of utility functions to be shared between frame buffer 2 and the new frame scheduling implementation based on frame buffer 3. Change-Id: Icc932c6c76fddeeedc8aa64ec27c7e0c955abfd0 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241604 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35743}
2021-12-15 15:05:15 +01:00
rtc_library("frame_helpers") {
sources = [
"frame_helpers.cc",
"frame_helpers.h",
]
deps = [
"../../api/video:encoded_frame",
"../../rtc_base:logging",
]
absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
}
New FrameBuffer3. FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe. Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35494}
2021-12-07 18:13:09 +01:00
rtc_library("frame_buffer") {
sources = [
"frame_buffer3.cc",
"frame_buffer3.h",
]
deps = [
":video_coding_utility",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
New FrameBuffer3. FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe. Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35494}
2021-12-07 18:13:09 +01:00
"../../api/units:timestamp",
"../../api/video:encoded_frame",
"../../rtc_base:logging",
"../../rtc_base:rtc_numerics",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional",
]
}
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
rtc_library("timing") {
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",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
"timing.cc",
"timing.h",
]
deps = [
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
"../../api/units:time_delta",
"../../api/video:video_rtp_headers",
Use Timestamp and TimeDelta in VCMTiming * Switches TimestampExtrapolator to use Timestamp as well. Bug: webrtc:13589 Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:13:55 +01:00
"../../rtc_base:logging",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
"../../rtc_base:macromagic",
"../../rtc_base:rtc_numerics",
"../../rtc_base/experiments:field_trial_parser",
"../../rtc_base/synchronization:mutex",
"../../rtc_base/time:timestamp_extrapolator",
"../../system_wrappers",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
Switch VCMRttFilter to use TimeDelta * Moved into its own GN target * Switched the internal buffer types to absl::InlinedVector as arrays are tricky to use with types that do not have default constructors. * Update fields arnd variables to use style guide. * Use constexpr for formerly const fields. * Adds unit tests. Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 15:22:08 +01:00
rtc_library("rtt_filter") {
sources = [
"rtt_filter.cc",
"rtt_filter.h",
]
deps = [ "../../api/units:time_delta" ]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
]
}
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
rtc_library("jitter_estimator") {
sources = [
"jitter_estimator.cc",
"jitter_estimator.h",
]
deps = [
":rtt_filter",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
"../../api/units:data_size",
"../../api/units:frequency",
"../../api/units:time_delta",
"../../api/units:timestamp",
"../../rtc_base",
"../../rtc_base:safe_conversions",
"../../rtc_base/experiments:jitter_upper_bound_experiment",
"../../system_wrappers",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
Migrate VCMInterFrameDelay to use Time units Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-07 14:50:51 +01:00
rtc_library("inter_frame_delay") {
sources = [
"inter_frame_delay.cc",
"inter_frame_delay.h",
]
deps = [
"..:module_api_public",
"../../api/units:frequency",
"../../api/units:time_delta",
"../../api/units:timestamp",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
rtc_library("video_coding") {
visibility = [ "*" ]
sources = [
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",
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",
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",
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",
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
"internal_defines.h",
Create LossNotificationController Create LossNotificationController, which produces LossNotification RTCP feedback messages when video packets/frames are lost. (LossNotification messages are sent when an RTP gap is detected, as well as when frames are later received which are undecodable because of the missing frames due to the previously dropped packets.) Bug: webrtc:10336 Change-Id: I7b3a156ed14e5a727349acdd82dae6997462421b Reviewed-on: https://webrtc-review.googlesource.com/c/123762 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26812}
2019-02-21 16:25:40 +01:00
"loss_notification_controller.cc",
"loss_notification_controller.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
"media_opt_util.cc",
"media_opt_util.h",
Break RtpFrameReferenceFinder into descriptor specific parts. This CL breaks out descriptor specific parts into separate classes. All logic in the newly added classes is just copy pasted from the (previously massive) RtpFrameReferenceFinder with the exception of how frames are being returned, which is now done via return value rather than a callback. Basically, all interesting changes have been made in the RtpFrameReferenceFinder. Bug: webrtc:12221 Change-Id: I5f958d2fbf4b77ba11c3c6c01d8d0d80e325be60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195448 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32717}
2020-11-27 17:56:37 +01:00
"rtp_frame_id_only_ref_finder.cc",
"rtp_frame_id_only_ref_finder.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",
Break RtpFrameReferenceFinder into descriptor specific parts. This CL breaks out descriptor specific parts into separate classes. All logic in the newly added classes is just copy pasted from the (previously massive) RtpFrameReferenceFinder with the exception of how frames are being returned, which is now done via return value rather than a callback. Basically, all interesting changes have been made in the RtpFrameReferenceFinder. Bug: webrtc:12221 Change-Id: I5f958d2fbf4b77ba11c3c6c01d8d0d80e325be60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195448 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32717}
2020-11-27 17:56:37 +01:00
"rtp_generic_ref_finder.cc",
"rtp_generic_ref_finder.h",
"rtp_seq_num_only_ref_finder.cc",
"rtp_seq_num_only_ref_finder.h",
"rtp_vp8_ref_finder.cc",
"rtp_vp8_ref_finder.h",
"rtp_vp9_ref_finder.cc",
"rtp_vp9_ref_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
"timestamp_map.cc",
"timestamp_map.h",
Split out counting unique rtp timestamps from packet_buffer Bug: None Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29656}
2019-10-30 14:12:24 +01:00
"unique_timestamp_counter.cc",
"unique_timestamp_counter.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",
New class VideoReceiver2, a trimmed-down vcm::VideoReceiver The vcm::VideoReceiver class is used by both VideoReceiveStream and the legacy api VideoCodingModule. They have different requirements, since the latter uses the old jitterbuffer and runs the code on a ProcessThread. By making a copy and trimming it down to what's actually used by VideoReceiveStream, we can drop the dependency on the old jitterbuffer, without breaking the legacy api. This should also make it easier to do follow-up refactorings to trim down the class further, and ultimately remove it. Bug: webrtc:7408 Change-Id: Iec8a167fe5d0425114b0b67a5b4c2fd5fc4fa150 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151910 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29108}
2019-09-09 12:51:55 +02:00
"video_receiver2.cc",
"video_receiver2.h",
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 av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +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",
New FrameBuffer3. FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe. Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35494}
2021-12-07 18:13:09 +01:00
":frame_buffer",
Add frame_helpers A number of utility functions to be shared between frame buffer 2 and the new frame scheduling implementation based on frame buffer 3. Change-Id: Icc932c6c76fddeeedc8aa64ec27c7e0c955abfd0 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241604 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35743}
2021-12-15 15:05:15 +01:00
":frame_helpers",
Migrate VCMInterFrameDelay to use Time units Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-07 14:50:51 +01:00
":inter_frame_delay",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
":jitter_estimator",
Switch VCMRttFilter to use TimeDelta * Moved into its own GN target * Switched the internal buffer types to absl::InlinedVector as arrays are tricky to use with types that do not have default constructors. * Update fields arnd variables to use style guide. * Use constexpr for formerly const fields. * Adds unit tests. Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 15:22:08 +01:00
":rtt_filter",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
":timing",
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",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"..:module_fec_api",
"../../api:array_view",
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",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Reland "Move api/rtp_headers.h to its own build target." This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e Original change's description: > Move api/rtp_headers.h to its own build target. > > Reduces dependencies on the libjingle_peerconnection_api target from > lower-level code. > > Bug: None > Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27213} Bug: None Tbr: kwiberg@webrtc.org Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-20 08:40:23 +01:00
"../../api:rtp_headers",
Fork VCMPacket for PacketBuffer into own struct it is easier to reduce and eliminate it when it is not bound to legacy video code Bug: webrtc:10979 Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-14 14:57:33 +01:00
"../../api:rtp_packet_info",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../../api:sequence_checker",
Add FrameBufferProxy in prep for FrameBuffer3 This is a delegate that is used by video_receive_stream2 to handle frame buffer tasks like threading, and stats. This will be used in a follow up to use FrameBuffer3 as a strategy selected by field trial. Unit-tests will be used in follow-up CLs containing Frame Buffer 3, and are expected to work with both Frame buffer proxy versions. Change-Id: I524279343d60a348d044d9085d618f12d7bf3a23 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241605 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35803}
2021-12-15 14:33:40 +01:00
"../../api/task_queue",
Refactor EncodeParameters usage, remove unused rtt/loss Bug: webrtc:10126 Change-Id: Ib93f5e65b25540576c026197f72a5902cf43fc16 Reviewed-on: https://webrtc-review.googlesource.com/c/114281 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26001}
2018-12-13 11:23:27 +01:00
"../../api/units:data_rate",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
"../../api/units:data_size",
"../../api/units:frequency",
Add field-trial parameter to enable tests simulating a slow decoder This CL adds a field trial parameter WebRTC-SlowDownDecoder that is used to simulate a slow decoder. The parameter specifies how many extra ms it takes to decode each video frame. This must only be used in manual testing. Bug: None Change-Id: Iad4079100d67b95c224277aaeaf572e38068717f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151911 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29153}
2019-09-11 12:00:22 +02:00
"../../api/units:time_delta",
Fix potential unsafe access to VCMTimestampMap::data The access to |_timestampMap| was guarded by a lock but not the access to the data pointer stored in |_timestampMap|. There was a potential race condition if new data was added in VCMGenericDecoder::Decode() while the data pointer retrieved from _timestampMap.Pop() was being used in VCMDecodedFrameCallback::Decoded(). This CL moves the storage of data to within |_timestampMap|, instead of being a pointer so that it's guarded by the same lock. Bug: webrtc:11229 Change-Id: I3f2afb568ed724db5719d508a73de402c4531dec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209361 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33374}
2021-03-03 14:39:44 +01:00
"../../api/units:timestamp",
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 10:02:56 -08:00
"../../api/video:builtin_video_bitrate_allocator_factory",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:encoded_frame",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../api/video:encoded_image",
Unify AdaptationReason and AdaptReason enums. Moves the unified AdaptationReason to the api/ folder. Bug: webrtc:11392 Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583 Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 11:36:55 +02:00
"../../api/video:video_adaptation",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../api/video:video_adaptation",
"../../api/video:video_bitrate_allocation",
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",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../api/video:video_bitrate_allocator_factory",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
Use AV1 packetizer/depacketizer for AV1 bitstreams Bug: webrtc:11042 Change-Id: Ibf45a99d8016dccbe109d946ac967efa927312e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161011 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29953}
2019-11-28 16:50:02 +01:00
"../../api/video:video_frame_type",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove 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 //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../../rtc_base:rtc_event",
Remove 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_numerics",
"../../rtc_base:rtc_task_queue",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../../rtc_base:threading",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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 field-trial parameter to enable tests simulating a slow decoder This CL adds a field trial parameter WebRTC-SlowDownDecoder that is used to simulate a slow decoder. The parameter specifies how many extra ms it takes to decode each video frame. This must only be used in manual testing. Bug: None Change-Id: Iad4079100d67b95c224277aaeaf572e38068717f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151911 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29153}
2019-09-11 12:00:22 +02:00
"../../rtc_base/experiments:field_trial_parser",
JitterEstimator: Remove old LowRate exp and add trial for upper bound. The frame time deltas are now capped based on the current noise. This has been tested in various conditions using both screen content and typical mobile video settings, to produce delays that are not overly high screen content, and simultaneously not negatively affect mobile calls on really bad network that may have high natural jitter. Bug: webrtc:9898 Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48 Reviewed-on: https://webrtc-review.googlesource.com/c/107349 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25469}
2018-11-01 11:20:49 +01:00
"../../rtc_base/experiments:jitter_upper_bound_experiment",
Fix WebRTC-Video-MinVideoBitrate for VP9 Make sure the experiment-derived value is used for VP9. Bug: webrtc:11024 Change-Id: I80b6d388486f2dec793bc8ca872babe6165dcfb2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156562 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29453}
2019-10-11 16:54:18 +02:00
"../../rtc_base/experiments:min_video_bitrate_experiment",
Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame. - Centralize field trial string reading to RateControlSettings - Cache RateControlSettings at all production code use sites Bug: None Change-Id: I0dbce9cc97fea0bc780982e7ef270b417a8c15bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158664 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29680}
2019-10-30 13:01:46 +01:00
"../../rtc_base/experiments:rate_control_settings",
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",
Introduces rtc_base/synchronization/mutex.h. This change introduces a new non-reentrant mutex to WebRTC. It enables eventual migration to Abseil's mutex. The mutex types supportable by webrtc::Mutex are - absl::Mutex - CriticalSection (Windows only) - pthread_mutex (POSIX only) In addition to introducing the mutexes, the CL also changes PacketBuffer to use the new mutex instead of rtc::CriticalSection. The method of yielding from critical_section.cc was given a mini-cleanup and YieldCurrentThread() was added to rtc_base/synchronization/yield.h/cc. Additionally, google_benchmark benchmarks for the mutexes were added (test courtesy of danilchap@), and some results from a pthread/Abseil shootout were added showing Abseil has the advantage in higher contention. Bug: webrtc:11567, webrtc:11634 Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31443}
2020-06-04 00:41:20 +02:00
"../../rtc_base/synchronization:mutex",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../rtc_base/system:no_unique_address",
Reland "Running FrameBuffer on task queue." This is a reland of 13943b7b7f6d00568912b9969db2c7871d18e21f Original change's description: > Running FrameBuffer on task queue. > > This prepares for running WebRTC in simulated time where event::Wait > based timing doesn't work. > > Bug: webrtc:10365 > Change-Id: Ia0f9b1cc8e3c8c27a38e45b40487050a4699d8cf > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129962 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27422} Bug: webrtc:10365 Change-Id: I412d3e0fe06c6dd57cdb42974f09e03f3a6ad038 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131124 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27572}
2019-04-11 12:39:34 +02:00
"../../rtc_base/task_utils:repeating_task",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../rtc_base/task_utils:to_queued_task",
Move base64.h to the proper location. Move base64.h to the proper location and put redirect header into the old place to be able to switch downstream users on new location. Bug: webrtc:8366 Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291 Reviewed-on: https://webrtc-review.googlesource.com/88223 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 17:38:12 +02:00
"../../rtc_base/third_party/base64",
Move 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",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
Use AV1 packetizer/depacketizer for AV1 bitstreams Bug: webrtc:11042 Change-Id: Ibf45a99d8016dccbe109d946ac967efa927312e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161011 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29953}
2019-11-28 16:50:02 +01:00
"../rtp_rtcp",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
Add av1 svc configuration for target bitrates This configuration mostly copies vp9 configuration for regular video, but is done separately to allow tune av1 svc bitrates independently of vp9. Bug: webrtc:12148 Change-Id: Icd11817ada8f9b6135ee2da57204eadb50de3954 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195329 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32713}
2020-11-25 17:01:22 +01:00
"../rtp_rtcp:rtp_video_header",
"codecs/av1:av1_svc_config",
]
absl_deps = [
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional",
"//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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 = [
Generic Frame Descriptor (GFD) VP8 templates. In this CL: - Updated Vp8TemporalLayers::OnEncodeDone to take a CodecSpecificInfo instead of a CodecSpecificInfoVP8, so that both the VP8 specific and generic information can be populated. - Added structs to represent the GFD template structure. - Added code to generate templates for video/screensharing. Bug: webrtc:10342 Change-Id: I978f9d708597a6f86bbdc494e62acf7a7b400db3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123422 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26987}
2019-03-04 16:37:50 +01:00
"include/video_codec_interface.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
"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 = [
Cleanup includes in modules/include/module_common_types.h Add missing includes to files that were transactivly depending on removed includes. Bug: None Change-Id: Id5923bb8dc3e1d8fbb664e460278ad3e5993be7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155963 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29396}
2019-10-07 12:56:24 +02:00
":codec_globals_headers",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_video",
"../../common_video/generic_frame_descriptor",
Export symbols needed by the Chromium component build (part 8). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 11:33:59 +02:00
"../../rtc_base/system:rtc_export",
Add 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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("video_coding_legacy") {
Add empty build target modules/video_coding:video_coding_legacy A followup cl will move VideoCodingModule and related code into this target. Bug: webrtc:7408 Change-Id: Iade572b597769456c9b8c76f584500e2bd9a58f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29122}
2019-09-10 08:29:17 +02:00
visibility = [ ":video_coding_unittests" ]
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
sources = [
"decoding_state.cc",
"decoding_state.h",
Move EventWrapper class to target video_coding_legacy. And remove some unneeded logic for WEBRTC_EVENT_INFINITE. Bug: webrtc:3380 Change-Id: Ibf632493edc6ced1609bd9ced44c2020fe9878cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169846 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30700}
2020-03-05 15:31:10 +01:00
"event_wrapper.cc",
"event_wrapper.h",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"frame_buffer.cc",
"frame_buffer.h",
"include/video_coding.h",
"jitter_buffer.cc",
"jitter_buffer.h",
Trim dependencies in modules/video_coding/ And move jitter_buffer_common.h to the legacy build target. Bug: None Change-Id: I986649f2f0773cadfa7dd9c8b533af7ecf01f3a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152382 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29150}
2019-09-10 15:28:17 +02:00
"jitter_buffer_common.h",
Fork VCMPacket for PacketBuffer into own struct it is easier to reduce and eliminate it when it is not bound to legacy video code Bug: webrtc:10979 Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-14 14:57:33 +01:00
"packet.cc",
"packet.h",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"receiver.cc",
"receiver.h",
"session_info.cc",
"session_info.h",
"video_coding_impl.cc",
"video_coding_impl.h",
"video_receiver.cc",
]
deps = [
":codec_globals_headers",
":encoded_frame",
Migrate VCMInterFrameDelay to use Time units Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-07 14:50:51 +01:00
":inter_frame_delay",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
":jitter_estimator",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
":timing",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
":video_codec_interface",
":video_coding",
"..:module_api",
"..:module_api_public",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../api:rtp_headers",
Fork VCMPacket for PacketBuffer into own struct it is easier to reduce and eliminate it when it is not bound to legacy video code Bug: webrtc:10979 Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-14 14:57:33 +01:00
"../../api:rtp_packet_info",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../../api:sequence_checker",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf This patch takes a stab at modules/video_coding, but reaches only about half. Bug: webrtc:10335 Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36335}
2022-03-25 12:43:14 +01:00
"../../api/transport:field_trial_based_config",
Use Timestamp to represent packet receive timestamps Before this CL, timestamps of received packets were rounded to the nearest millisecond and stored as int64_t. Due to the rounding it sometimes happened that timestamps later in the pipeline that are not rounded seem to occur even before the video frame was received. Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18 Bug: webrtc:12722 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398 Reviewed-by: Chen Xing <chxg@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33916}
2021-04-30 13:10:56 +02:00
"../../api/units:timestamp",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../api/video:encoded_image",
"../../api/video:video_frame",
Fork VCMPacket for PacketBuffer into own struct it is easier to reduce and eliminate it when it is not bound to legacy video code Bug: webrtc:10979 Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-14 14:57:33 +01:00
"../../api/video:video_frame_type",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../api/video:video_rtp_headers",
"../../api/video_codecs:video_codecs_api",
"../../common_video",
Include module_common_types.h only where needed Bug: None Change-Id: I73d493f8f186b429c7be808f4dfac0398f150931 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153891 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29277}
2019-09-23 10:31:16 +02:00
"../../modules/rtp_rtcp:rtp_video_header",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../rtc_base:checks",
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../rtc_base:rtc_base_approved",
Move EventWrapper class to target video_coding_legacy. And remove some unneeded logic for WEBRTC_EVENT_INFINITE. Bug: webrtc:3380 Change-Id: Ibf632493edc6ced1609bd9ced44c2020fe9878cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169846 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30700}
2020-03-05 15:31:10 +01:00
"../../rtc_base:rtc_event",
Remove //rtc_base:safe_conversions from public deps Bug: webrtc:8603 Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36452}
2022-04-04 17:04:37 +02:00
"../../rtc_base:safe_conversions",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf This patch takes a stab at modules/video_coding, but reaches only about half. Bug: webrtc:10335 Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36335}
2022-03-25 12:43:14 +01:00
"../../rtc_base/memory:always_valid_pointer",
Migrate modules/video_coding to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 12:17:12 +02:00
"../../rtc_base/synchronization:mutex",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../../system_wrappers",
Fork VCMPacket for PacketBuffer into own struct it is easier to reduce and eliminate it when it is not bound to legacy video code Bug: webrtc:10979 Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-14 14:57:33 +01:00
"../rtp_rtcp:rtp_rtcp_format",
"../rtp_rtcp:rtp_video_header",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"../utility",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
"//third_party/abseil-cpp/absl/types:variant",
]
Add empty build target modules/video_coding:video_coding_legacy A followup cl will move VideoCodingModule and related code into this target. Bug: webrtc:7408 Change-Id: Iade572b597769456c9b8c76f584500e2bd9a58f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29122}
2019-09-10 08:29:17 +02:00
}
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("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
Delete header file mock_vcm_callbacks.h Move definitions of mock classes to the only user, the unit tests for the deprecated class vcm::VideoReceiver. Bug: webrtc:7408 Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30451}
2020-02-04 10:46:37 +01:00
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 = [
video: Implement bandwidth based scaler The |slice_qp_detla| reported by the hardware is not credible, which causing the quality scaler cannot work properly,the resolution cannot be adjusted correctly. To fix this issue, this CL implements a bandwidth scaler which is used for adjust resolution, this scaler will be used when QP based quality scaler is not working due to untrusted QP reported by HW AVC encoder. Bug: webrtc:12942 Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 17:19:44 +08:00
"utility/bandwidth_quality_scaler.cc",
"utility/bandwidth_quality_scaler.h",
Introduce DecodedFramesHistory class and use it in FrameBuffer This is a space efficient way to store more records about decoded frames, which is needed for long term references. Bug: webrtc:9710 Change-Id: I051d59d34a966d48db011142466d9cd15304b7d9 Reviewed-on: https://webrtc-review.googlesource.com/c/116792 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26240}
2019-01-14 13:24:22 +01:00
"utility/decoded_frames_history.cc",
"utility/decoded_frames_history.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 frame drop functionality in VideoAdapter into a separate class. This class will replace modules/video_coding/utility/framerate_controller.h Bug: webrtc:13031 Change-Id: I8faa9c3c158b7c5ab0618e3504224c7e00f8e0b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227350 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34802}
2021-08-16 09:33:13 +02:00
"utility/framerate_controller_deprecated.cc",
"utility/framerate_controller_deprecated.h",
Disambiguate kIvfHeaderSize for Mozilla unified build kIvfHeaderSize is defined both inside of ivf_file_writer.cc and ivf_file_reader.cc. This patch moves its definition into a header. Bug: webrtc:13463 Change-Id: Ia6b2fcc3434f69a1e30a7dae7bf0c90547f11d98 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239722 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35540}
2021-12-08 12:07:09 -08:00
"utility/ivf_defines.h",
Introduce IVF file reader Bug: webrtc:10138 Change-Id: I97d332942f4e645527330159efefb1cb1d8034a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160008 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29844}
2019-11-20 13:30:19 +01:00
"utility/ivf_file_reader.cc",
"utility/ivf_file_reader.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",
Reland "Parse encoded frame QP if not provided by encoder" This reverts commit 727d2afc4330efebc904e0e4f366e885d7b08787. Reason for revert: Use thread-safe wrapper for H264 parser. Original change's description: > Revert "Parse encoded frame QP if not provided by encoder" > > This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508. > > Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview > > Original change's description: > > Parse encoded frame QP if not provided by encoder > > > > Bug: webrtc:12542 > > Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33434} > > TBR=asapersson@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12542 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33441} # Not skipping CQ checks because this is a reland. Bug: webrtc:12542 Change-Id: Ib7601fd6f2f26bceddbea2b4ba54d67a281f3a59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211660 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33458}
2021-03-15 10:12:57 +01:00
"utility/qp_parser.cc",
"utility/qp_parser.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",
Reland "Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields." This is a reland of 3097008de03b6260da5cfabb5cbac6f6a64ca810 Patchset 1 is a pure reland. Patchset 2 contains a bugfix plus a test covering that case. Bug: webrtc:12354, chromium:1230448 Original change's description: > Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields. > > These fields will be used for bitstream validation in upcoming CLs. > A new vp9_constants.h file is also added, containing common constants > defined by the bitstream spec. > > Bug: webrtc:12354 > Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34476} Bug: webrtc:12354 Change-Id: Ibd301eb458a6104b562cefbc0e616c39b54fb38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229060 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34789}
2021-08-17 19:58:52 +02:00
"utility/vp9_constants.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
]
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 "Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields." This is a reland of 3097008de03b6260da5cfabb5cbac6f6a64ca810 Patchset 1 is a pure reland. Patchset 2 contains a bugfix plus a test covering that case. Bug: webrtc:12354, chromium:1230448 Original change's description: > Rename vp9::FrameInfo to vp9::UncompressedHeader and add more fields. > > These fields will be used for bitstream validation in upcoming CLs. > A new vp9_constants.h file is also added, containing common constants > defined by the bitstream spec. > > Bug: webrtc:12354 > Change-Id: If04256d83409069c8bee43ad41aed41c3707dfd3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226060 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34476} Bug: webrtc:12354 Change-Id: Ibd301eb458a6104b562cefbc0e616c39b54fb38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229060 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34789}
2021-08-17 19:58:52 +02:00
"../../api:array_view",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Asynchronous QualityScaler: Callback-based CheckQpTask. This CL breaks up the CheckQp() operation into several steps managed by the inner helper class CheckQpTask, making responding to high or low QP an asynchronous operation. Why? Reconfiguring the stream in response to QP overuse will in the future be handled on a separate task queue. See Call-Level Adaptation Processing for more details: https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing Instead of "bool AdaptDown()" when high QP is reported, synchronously returning true or false depending on the result of adaptation, this CL introduces void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh( rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>); Where QualityScalerQpUsageHandlerCallback::OnQpUsageHandled( bool clear_qp_samples); Instructs the QualityScaler whether to clear samples before checking QP the next time or to increase the frequency of checking (corresponding to AdaptDown's return value prior to this CL). QualityScaler no longer using AdaptationObserverInterface, this class is renamed and moved to overuse_frame_detector.h. The dependency between CheckQpTasks is made explicit with CheckQpTask::Result and variables like observed_enough_frames_, adapt_called_ and adapt_failed_ are moved there and given more descriptive names. Bug: webrtc:11521 Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-27 17:40:55 +02:00
"../../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../../api:sequence_checker",
Introduce DecodedFramesHistory class and use it in FrameBuffer This is a space efficient way to store more records about decoded frames, which is needed for long term references. Bug: webrtc:9710 Change-Id: I051d59d34a966d48db011142466d9cd15304b7d9 Reviewed-on: https://webrtc-review.googlesource.com/c/116792 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26240}
2019-01-14 13:24:22 +01:00
"../../api/video:encoded_frame",
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",
Unify AdaptationReason and AdaptReason enums. Moves the unified AdaptationReason to the api/ folder. Bug: webrtc:11392 Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583 Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 11:36:55 +02:00
"../../api/video:video_adaptation",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_bitrate_allocation",
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 "Parse encoded frame QP if not provided by encoder" This reverts commit 727d2afc4330efebc904e0e4f366e885d7b08787. Reason for revert: Use thread-safe wrapper for H264 parser. Original change's description: > Revert "Parse encoded frame QP if not provided by encoder" > > This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508. > > Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview > > Original change's description: > > Parse encoded frame QP if not provided by encoder > > > > Bug: webrtc:12542 > > Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33434} > > TBR=asapersson@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12542 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33441} # Not skipping CQ checks because this is a reland. Bug: webrtc:12542 Change-Id: Ib7601fd6f2f26bceddbea2b4ba54d67a281f3a59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211660 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33458}
2021-03-15 10:12:57 +01:00
"../../api/video:video_codec_constants",
Removed unused include from encoded_image.h. Bug: webrtc:9378 Change-Id: Ie26ab4d30d62ec109a8be638661789399821c162 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179525 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31758}
2020-07-17 15:19:40 +02:00
"../../api/video:video_frame",
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",
Change ParseUncompressedVp9Header implementation to use BitstreamReader Bug: None Change-Id: I91010b0102622fd8154f8ba941e61298b0584eae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230802 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34899}
2021-09-01 16:27:04 +02:00
"../../rtc_base:bitstream_reader",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../../rtc_base:refcount",
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",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Asynchronous QualityScaler: Callback-based CheckQpTask. This CL breaks up the CheckQp() operation into several steps managed by the inner helper class CheckQpTask, making responding to high or low QP an asynchronous operation. Why? Reconfiguring the stream in response to QP overuse will in the future be handled on a separate task queue. See Call-Level Adaptation Processing for more details: https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing Instead of "bool AdaptDown()" when high QP is reported, synchronously returning true or false depending on the result of adaptation, this CL introduces void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh( rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>); Where QualityScalerQpUsageHandlerCallback::OnQpUsageHandled( bool clear_qp_samples); Instructs the QualityScaler whether to clear samples before checking QP the next time or to increase the frequency of checking (corresponding to AdaptDown's return value prior to this CL). QualityScaler no longer using AdaptationObserverInterface, this class is renamed and moved to overuse_frame_detector.h. The dependency between CheckQpTasks is made explicit with CheckQpTask::Result and variables like observed_enough_frames_, adapt_called_ and adapt_failed_ are moved there and given more descriptive names. Bug: webrtc:11521 Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-27 17:40:55 +02:00
"../../rtc_base:weak_ptr",
video: Implement bandwidth based scaler The |slice_qp_detla| reported by the hardware is not credible, which causing the quality scaler cannot work properly,the resolution cannot be adjusted correctly. To fix this issue, this CL implements a bandwidth scaler which is used for adjust resolution, this scaler will be used when QP based quality scaler is not working due to untrusted QP reported by HW AVC encoder. Bug: webrtc:12942 Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 17:19:44 +08:00
"../../rtc_base/experiments:bandwidth_quality_scaler_settings",
"../../rtc_base/experiments:encoder_info_settings",
Add ability to configure quality scaler settings through field trial. optional<int> min_frames: The minimum number frames to observe to make a scaling decision. Default: kMinFramesNeededToScale in quality_scaler.cc optional<double> initial_scale_factor: The sample period scale factor. Default: kSamplePeriodScaleFactor in quality_scaler.cc optional<double> scale_factor: Option to use a reduced sampling interval when last check did not result in an adaptation (if unset the initial_scale_factor is used). Bug: none Change-Id: I3bb955d1f8d7d7d49bc118361614b5aa59605231 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135125 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27860}
2019-05-06 14:17:35 +02:00
"../../rtc_base/experiments:quality_scaler_settings",
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",
Move simulcast hysteresis factor parsing to RateControlSettings Bug: webrtc:10223 Change-Id: I962ca959afbcd8c27a0f79533c6e3c97369c697e Reviewed-on: https://webrtc-review.googlesource.com/c/119262 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26374}
2019-01-23 16:53:18 +01:00
"../../rtc_base/experiments:rate_control_settings",
Implement stable rate support in SimulcastRateAllocator Bug: webrtc:10126 Change-Id: I2ea8d27b0bd6f7ffd1ebbba451bd1ce1f2eee3d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151121 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29097}
2019-09-04 16:30:47 +02:00
"../../rtc_base/experiments:stable_target_rate_experiment",
Reland "Parse encoded frame QP if not provided by encoder" This reverts commit 727d2afc4330efebc904e0e4f366e885d7b08787. Reason for revert: Use thread-safe wrapper for H264 parser. Original change's description: > Revert "Parse encoded frame QP if not provided by encoder" > > This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508. > > Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview > > Original change's description: > > Parse encoded frame QP if not provided by encoder > > > > Bug: webrtc:12542 > > Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33434} > > TBR=asapersson@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12542 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33441} # Not skipping CQ checks because this is a reland. Bug: webrtc:12542 Change-Id: Ib7601fd6f2f26bceddbea2b4ba54d67a281f3a59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211660 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33458}
2021-03-15 10:12:57 +01:00
"../../rtc_base/synchronization:mutex",
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",
Delete rtc::File, usage replaced with FileWrapper Bug: webrtc:6463 Change-Id: Ia0767a2e6bbacc43e63c30ed3bd3edb10ff6e645 Reviewed-on: https://webrtc-review.googlesource.com/c/121943 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26613}
2019-02-08 16:40:53 +01:00
"../../rtc_base/system:file_wrapper",
Reland "Introduce RTC_NO_UNIQUE_ADDRESS." This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447 This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since there have been some issues. Original change's description: > Introduce RTC_NO_UNIQUE_ADDRESS. > > This macro introduces the possibility to suggest the compiler that a > data member doesn't need an address different from other non static > data members. > > The usage of a macro is to maintain portability since at the moment > the attribute [[no_unique_address]] is only supported by clang > with at least -std=c++11 but it should be supported by all the > compilers starting from C++20. > > Bug: webrtc:11495 > Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32246} Bug: webrtc:11495, webrtc:12218 Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:07:42 +01:00
"../../rtc_base/system:no_unique_address",
Adds repeating task class. This CL adds a single class to manage the use case of having a task that repeats itself by a fixed or variable interval. It replaces the repeating task previously locally defined for rtp transport controller send as well as the cancelable periodic task. Furthermore, it is introduced where one off repeating tasks were created before. It provides the currently used functionality of the cancelable periodic task, but not some of the unused features, such as allowing cancellation of tasks before they are started and cancellation of a task after the owning task queue has been destroyed. Bug: webrtc:9883 Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca Reviewed-on: https://webrtc-review.googlesource.com/c/116182 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26313}
2019-01-18 10:30:54 +01:00
"../../rtc_base/task_utils:repeating_task",
Asynchronous QualityScaler: Callback-based CheckQpTask. This CL breaks up the CheckQp() operation into several steps managed by the inner helper class CheckQpTask, making responding to high or low QP an asynchronous operation. Why? Reconfiguring the stream in response to QP overuse will in the future be handled on a separate task queue. See Call-Level Adaptation Processing for more details: https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing Instead of "bool AdaptDown()" when high QP is reported, synchronously returning true or false depending on the result of adaptation, this CL introduces void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh( rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>); Where QualityScalerQpUsageHandlerCallback::OnQpUsageHandled( bool clear_qp_samples); Instructs the QualityScaler whether to clear samples before checking QP the next time or to increase the frequency of checking (corresponding to AdaptDown's return value prior to this CL). QualityScaler no longer using AdaptationObserverInterface, this class is renamed and moved to overuse_frame_detector.h. The dependency between CheckQpTasks is made explicit with CheckQpTask::Result and variables like observed_enough_frames_, adapt_called_ and adapt_failed_ are moved there and given more descriptive names. Bug: webrtc:11521 Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-27 17:40:55 +02:00
"../../rtc_base/task_utils:to_queued_task",
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",
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
]
Refactors Vp9UncompressedHeaderParser. Biggest change is a new helper class used to read data from the bitstream and then pass the result to a function if reading was successful. There's also helper to do if/else flow based on the read values. This avoids a bunch of temporaries and in my view makes the code esaier to read. For example, this block: uint32_t bit; RETURN_FALSE_IF_ERROR(br->ReadBits(&bit, 1)); if (bit) { RETURN_FALSE_IF_ERROR(br->ConsumeBits(7)); } ...is now written as: RETURN_IF_FALSE( br->IfNextBoolean([br] { return br->ConsumeBits(7); })); In addition, we parse and put a few extra things in FrameInfo: show_existing_frame, is_keyframe, and base_qp. Bug: webrtc:12354 Change-Id: Ia0b707b223a1afe0a4521ce2b995437d41243c06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215239 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33776}
2021-04-19 11:46:12 +02:00
absl_deps = [
Change ParseUncompressedVp9Header implementation to use BitstreamReader Bug: None Change-Id: I91010b0102622fd8154f8ba941e61298b0584eae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230802 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34899}
2021-09-01 16:27:04 +02:00
"//third_party/abseil-cpp/absl/numeric:bits",
Refactors Vp9UncompressedHeaderParser. Biggest change is a new helper class used to read data from the bitstream and then pass the result to a function if reading was successful. There's also helper to do if/else flow based on the read values. This avoids a bunch of temporaries and in my view makes the code esaier to read. For example, this block: uint32_t bit; RETURN_FALSE_IF_ERROR(br->ReadBits(&bit, 1)); if (bit) { RETURN_FALSE_IF_ERROR(br->ConsumeBits(7)); } ...is now written as: RETURN_IF_FALSE( br->IfNextBoolean([br] { return br->ConsumeBits(7); })); In addition, we parse and put a few extra things in FrameInfo: show_existing_frame, is_keyframe, and base_qp. Bug: webrtc:12354 Change-Id: Ia0b707b223a1afe0a4521ce2b995437d41243c06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215239 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33776}
2021-04-19 11:46:12 +02:00
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
Use preprocessor to strip H264 implementation. This CL makes it more flexible and easier to include/exclude H264 code when using other build systems because it delegates the decision to remove the code to the preprocessor instead of GN. This CL should be a noop, and for WebRTC/Chromium the GN param `rtc_use_h264` will still be the only thing to change in order to include/exclude H264. Moving code that requires ffmpeg or h264 out of the #ifdef/#endif part should break the build since dependencies are only added if `rtc_use_h264=true`. Bug: webrtc:9213 Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28055}
2019-05-24 12:17:59 +02:00
"codecs/h264/h264_color_space.cc",
"codecs/h264/h264_color_space.h",
"codecs/h264/h264_decoder_impl.cc",
"codecs/h264/h264_decoder_impl.h",
"codecs/h264/h264_encoder_impl.cc",
"codecs/h264/h264_encoder_impl.h",
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
"codecs/h264/include/h264.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
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",
Add fix for 8-bit H264 HDR content 8-bit H264 HDR content is not rendered correctly in Chrome on Windows. This is a temporary fix that converts the 8-bit buffer to a 10-bit buffer if the color space indicates that the buffer should be rendered as HDR. Bug: webrtc:10575 Change-Id: I106612ec489c6371fa774424a4cf07d9bad40fc3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134040 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27766}
2019-04-25 11:51:55 +02:00
"../../api/video:video_frame_i010",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Use preprocessor to strip H264 implementation. This CL makes it more flexible and easier to include/exclude H264 code when using other build systems because it delegates the decision to remove the code to the preprocessor instead of GN. This CL should be a noop, and for WebRTC/Chromium the GN param `rtc_use_h264` will still be the only thing to change in order to include/exclude H264. Moving code that requires ffmpeg or h264 out of the #ifdef/#endif part should break the build since dependencies are only added if `rtc_use_h264=true`. Bug: webrtc:9213 Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28055}
2019-05-24 12:17:59 +02:00
"../../common_video",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Export symbols needed by the Chromium component build (part 3). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). Bug: webrtc:9419 Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94 Reviewed-on: https://webrtc-review.googlesource.com/c/106343 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 14:13:50 +02:00
"../../rtc_base/system:rtc_export",
Add fix for 8-bit H264 HDR content 8-bit H264 HDR content is not rendered correctly in Chrome on Windows. This is a temporary fix that converts the 8-bit buffer to a 10-bit buffer if the color space indicates that the buffer should be rendered as HDR. Bug: webrtc:10575 Change-Id: I106612ec489c6371fa774424a4cf07d9bad40fc3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134040 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27766}
2019-04-25 11:51:55 +02:00
"../../system_wrappers:field_trial",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../system_wrappers:metrics",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
"//third_party/libyuv",
]
absl_deps = [
Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" This is a reland of 80cd25bcfb2264fa0f1192de942a6f063879dd42 Original change's description: > Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase > > Bug: None > Change-Id: I225fe1e16a3c96e5a03e3ae8fe975f368be7e6ad > Reviewed-on: https://webrtc-review.googlesource.com/c/107303 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25312} Tbr: kwiberg@webrtc.org Bug: None Change-Id: Id43a93bada9d6d66a4d0f0286f583066156aa2fc Reviewed-on: https://webrtc-review.googlesource.com/c/107716 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25368}
2018-10-23 10:07:25 +02:00
"//third_party/abseil-cpp/absl/strings",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"//third_party/abseil-cpp/absl/types:optional",
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) {
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 += [
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"//third_party/ffmpeg",
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/openh264:encoder",
]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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
]
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",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../../api:fec_controller_api",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Move 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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
Add 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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_video",
Signal requested resolution alignment requirements from sinks to sources. Bug: webrtc:11218 Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 09:47:11 +01:00
"../../media:rtc_media_base",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Migrate modules/video_coding to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 12:17:12 +02:00
"../../rtc_base/synchronization:mutex",
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
]
}
Splits vp9_impl into libvpx_vp9_encoder and libvpx_vp9_decoder. Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and drops vp8 from the name. Follow-up CLs will wire up actual usage in the new classes through the interface so that we can unit test them more easily. Bug: webrtc:12274 Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32816}
2020-12-09 20:58:21 +01:00
# This target defines a bare-bones interface towards libvpx, used by the
# VP8 and VP9 wrappers below.
rtc_library("webrtc_libvpx_interface") {
visibility = [ "*" ]
sources = [
"codecs/interface/libvpx_interface.cc",
"codecs/interface/libvpx_interface.h",
]
deps = [ "../../rtc_base:checks" ]
if (rtc_build_libvpx) {
deps += [ rtc_libvpx_dir ]
}
}
Add fuzzer to validate libvpx vp9 encoder wrapper Fix simulcast svc controller to reuse dropped frame configuration, same as full svc and k-svc controllers do. This fuzzer reminded the issue was still there. This is a reland of https://webrtc-review.googlesource.com/c/src/+/212281 Bug: webrtc:11999 Change-Id: Id3b2cd6c7e0923adfffb4e04c35ed2d6faca6704 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215921 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33802}
2021-04-21 14:35:54 +02:00
rtc_library("mock_libvpx_interface") {
testonly = true
sources = [ "codecs/interface/mock_libvpx_interface.h" ]
deps = [
":webrtc_libvpx_interface",
"../../test:test_support",
]
}
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.
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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 = [
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 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",
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
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",
Splits vp9_impl into libvpx_vp9_encoder and libvpx_vp9_decoder. Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and drops vp8 from the name. Follow-up CLs will wire up actual usage in the new classes through the interface so that we can unit test them more easily. Bug: webrtc:12274 Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32816}
2020-12-09 20:58:21 +01:00
":webrtc_libvpx_interface",
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
":webrtc_vp8_temporal_layers",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../../api:fec_controller_api",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Move 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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Use single FrameBufferController in VP8, created by a factory. This CL paves the way to making FrameBufferController injectable. LibvpxVp8Encoder can manage multiple streams. Prior to this CL, each stream had its own frame buffer controller, all of them held in a vector by LibvpxVp8Encoder. This complicated the code and produced some code duplication (cf. SetupTemporalLayers). This CL: 1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs will make this factory injectable.) 2. Makes LibvpxVp8Encoder use a single controller. This single controller will, in the case of multiple streams, delegate its work to multiple controllers, but that fact is not visible to LibvpxVp8Encoder. This CL also squashes CL #126046 (Send notifications of RTT and PLR changes to Vp8FrameBufferController) into it. Bug: webrtc:10382 Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27206}
2019-03-20 11:56:20 +01:00
"../../api/video_codecs:vp8_temporal_layers_factory",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove 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",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Add field trial for vp8 cpu speed configuration for arm. Bug: none Change-Id: I90c2475a229a1f10016e2ad84029e19b5a4f9966 Reviewed-on: https://webrtc-review.googlesource.com/c/107340 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25484}
2018-11-02 11:38:46 +01:00
"../../rtc_base/experiments:cpu_speed_experiment",
LibvpxVp8Encoder: add option to configure resolution_bitrate_limits. Bug: none Change-Id: Ia01d630fc95e19a4a08cd7a004238c22d823b4dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205521 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33144}
2021-02-03 10:16:30 +01:00
"../../rtc_base/experiments:encoder_info_settings",
Create field trial for vp8 number of thread on iOS. Without the added preprocessor check, iOS device will be using the desktop logic to determine the number of thread. This put iPhone 8 and iPhone X to use 3 threads and all other iPhones after iPhone 5 to use a single thread. This CL added a preprocessor for WEBRTC_IOS to have it own thread number calculation logic. In which, the maximum number of thread is fetched from a field_trial and capped by the number of CPU available on the device. Bug: webrtc:10005 Change-Id: I8c6257fcbf85b07bc986b5f733dbabb3feee37f7 Reviewed-on: https://webrtc-review.googlesource.com/c/110941 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25997}
2018-12-11 14:25:25 -08:00
"../../rtc_base/experiments:field_trial_parser",
Move parsing of trusted rate controller to RateControlSettings Bug: webrtc:10223 Change-Id: Iadf46e278e0f994ed95ff1a240c1f39a0421ab7c Reviewed-on: https://webrtc-review.googlesource.com/c/119261 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26371}
2019-01-23 12:48:13 +01:00
"../../rtc_base/experiments:rate_control_settings",
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",
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
]
Support native scaling of VideoFrameBuffers in LibvpxVp8Encoder. This is a follow-up to the VP9, fixing VP8 this time. Context again: This CL is part of Optimized Scaling efforts. In Chromium, the native frame buffer is getting an optimized CropAndScale() implementation. To support HW accelerated scaling, returning pre-scaled images and skipping unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale instead of libyuv::XXXXScale and 2) only map buffers it actually intends to encode. - To achieve this, WebRTC encoders are updated to map kNative video buffers so that in a follow-up CL VideoStreamEncoder can stop mapping intermediate buffer sizes. Bug: webrtc:12469, chromium:1157072 Change-Id: I026527ae77e36f66d02e149ad6fe304f6a8ccb05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212600 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#33537}
2021-03-23 09:18:28 +01:00
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
Reland "Handle scalability mode in QueryCodecSupport" This reverts commit 74281bed5350af9c15f83e0b1aec5c5921dbf76f. Reason for revert: Fixed unit test by removing VP9 profile 2 from encoder factory unit test since this is platform dependent. Original change's description: > Revert "Handle scalability mode in QueryCodecSupport" > > This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5. > > Reason for revert: Speculative revert. Breaks upstream project http://b/200009579 > > Original change's description: > > Handle scalability mode in QueryCodecSupport > > > > All valid scalability modes should be supported by the builtin > > software decoder/encoder. > > > > Bug: chromium:1187565 > > Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34998} > > TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1187565 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35001} Bug: chromium:1187565 Change-Id: I598a2a530b8fea22997bbb5910eb3b864d1e28a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232021 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35003}
2021-09-15 10:56:04 +00:00
"//third_party/abseil-cpp/absl/strings:strings",
Support native scaling of VideoFrameBuffers in LibvpxVp8Encoder. This is a follow-up to the VP9, fixing VP8 this time. Context again: This CL is part of Optimized Scaling efforts. In Chromium, the native frame buffer is getting an optimized CropAndScale() implementation. To support HW accelerated scaling, returning pre-scaled images and skipping unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale instead of libyuv::XXXXScale and 2) only map buffers it actually intends to encode. - To achieve this, WebRTC encoders are updated to map kNative video buffers so that in a follow-up CL VideoStreamEncoder can stop mapping intermediate buffer sizes. Bug: webrtc:12469, chromium:1157072 Change-Id: I026527ae77e36f66d02e149ad6fe304f6a8ccb05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212600 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#33537}
2021-03-23 09:18:28 +01:00
"//third_party/abseil-cpp/absl/types:optional",
]
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
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("webrtc_vp8_temporal_layers") {
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
visibility = [ "*" ]
sources = [
"codecs/vp8/default_temporal_layers.cc",
"codecs/vp8/default_temporal_layers.h",
"codecs/vp8/include/temporal_layers_checker.h",
"codecs/vp8/screenshare_layers.cc",
"codecs/vp8/screenshare_layers.h",
"codecs/vp8/temporal_layers.h",
"codecs/vp8/temporal_layers_checker.cc",
]
deps = [
":codec_globals_headers",
":video_codec_interface",
":video_coding_utility",
Pass FecControllerOverride to Vp8FrameBufferController Bug: webrtc:10769 Change-Id: I06d875f5afdc7ebf290ad70934b6632e20ddf065 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143964 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28418}
2019-06-28 18:12:15 +02:00
"../../api:fec_controller_api",
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
"../../api/video_codecs:video_codecs_api",
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
}
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.
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("webrtc_vp9_helpers") {
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
sources = [
"codecs/vp9/svc_config.cc",
"codecs/vp9/svc_config.h",
]
deps = [
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
":codec_globals_headers",
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
":video_codec_interface",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_bitrate_allocation",
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",
Add support for stable bitrate target in SvcRateAllocator Bug: webrtc:10126 Change-Id: I1362d183bb91510db4e2763a779bcdf681d855ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149069 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29066}
2019-09-04 14:30:57 +02:00
"../../api/video:video_codec_constants",
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",
Allow HVGA Vp9 SVC to have 2 spatial layers and remove excessive rounding Bug: webrtc:11652 Change-Id: I8bfa91c3115d6ebb17beefbb2a5e51efbbd599e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177000 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31502}
2020-06-11 14:01:56 +02:00
"../../rtc_base:logging",
Add support for stable bitrate target in SvcRateAllocator Bug: webrtc:10126 Change-Id: I1362d183bb91510db4e2763a779bcdf681d855ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149069 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29066}
2019-09-04 14:30:57 +02:00
"../../rtc_base/experiments:stable_target_rate_experiment",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/container:inlined_vector" ]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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" ]
Always compile VP9 source files. Instead of optionally compile VP9 source files based on the value of the GN argument 'rtc_libvpx_build_vp9', this CL uses the preprocessor macro RTC_ENABLE_VP9 to decide if VP9 related code needs to be compiled or not. Bug: None Change-Id: I5c1b69d7ec35e8446181d98c912277d0ae8fdba2 Reviewed-on: https://webrtc-review.googlesource.com/c/111063 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25685}
2018-11-19 09:52:37 +01:00
sources = [
"codecs/vp9/include/vp9.h",
Splits vp9_impl into libvpx_vp9_encoder and libvpx_vp9_decoder. Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and drops vp8 from the name. Follow-up CLs will wire up actual usage in the new classes through the interface so that we can unit test them more easily. Bug: webrtc:12274 Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32816}
2020-12-09 20:58:21 +01:00
"codecs/vp9/libvpx_vp9_decoder.cc",
"codecs/vp9/libvpx_vp9_decoder.h",
"codecs/vp9/libvpx_vp9_encoder.cc",
"codecs/vp9/libvpx_vp9_encoder.h",
Always compile VP9 source files. Instead of optionally compile VP9 source files based on the value of the GN argument 'rtc_libvpx_build_vp9', this CL uses the preprocessor macro RTC_ENABLE_VP9 to decide if VP9 related code needs to be compiled or not. Bug: None Change-Id: I5c1b69d7ec35e8446181d98c912277d0ae8fdba2 Reviewed-on: https://webrtc-review.googlesource.com/c/111063 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25685}
2018-11-19 09:52:37 +01:00
"codecs/vp9/vp9.cc",
"codecs/vp9/vp9_frame_buffer_pool.cc",
"codecs/vp9/vp9_frame_buffer_pool.h",
]
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",
Splits vp9_impl into libvpx_vp9_encoder and libvpx_vp9_decoder. Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and drops vp8 from the name. Follow-up CLs will wire up actual usage in the new classes through the interface so that we can unit test them more easily. Bug: webrtc:12274 Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32816}
2020-12-09 20:58:21 +01:00
":webrtc_libvpx_interface",
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",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../../api:fec_controller_api",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf rename WebRtcKeyValueConfig to FieldTrialsView Bug: webrtc:10335 Change-Id: If725bd498c4c3daf144bee638230fa089fdde833 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 11:04:48 +02:00
"../../api:field_trials_view",
Make RefCountedObject require overriding virtual methods Bug: webrtc:12701 Change-Id: Ia4ae4ad2e857cb8790d6ccfb6f88f07d52a8e91b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215967 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33831}
2021-04-26 11:46:54 +02:00
"../../api:refcountedbase",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Adds field trial to set per-layer speed for libvpx vp9. The trial name WebRTC-VP9-PerLayerSpeed is used to a) set encoding speed per spatial layer, based on resolution b) allow explicitly overriding speed per layer, for testing Additionally, this CL updates the vp9 wrapper in preparation for injectable trials. Bug: webrtc:11551, webrtc:11926 Change-Id: I2bb3a664feaef60483ffc241b71070284d3e0172 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186400 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32294}
2020-10-01 19:13:13 +02:00
"../../api/transport:field_trial_based_config",
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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
Add 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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
LibvpxVp9Encoder: add option to configure resolution_bitrate_limits. Bug: none Change-Id: Icdd7333296d652b1e0c159226df702084303475c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204701 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33121}
2021-02-01 09:20:05 +01:00
"../../rtc_base/experiments:encoder_info_settings",
Adds field trial to set per-layer speed for libvpx vp9. The trial name WebRTC-VP9-PerLayerSpeed is used to a) set encoding speed per spatial layer, based on resolution b) allow explicitly overriding speed per layer, for testing Additionally, this CL updates the vp9 wrapper in preparation for injectable trials. Bug: webrtc:11551, webrtc:11926 Change-Id: I2bb3a664feaef60483ffc241b71070284d3e0172 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186400 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32294}
2020-10-01 19:13:13 +02:00
"../../rtc_base/experiments:field_trial_parser",
Move parsing of trusted rate controller to RateControlSettings Bug: webrtc:10223 Change-Id: Iadf46e278e0f994ed95ff1a240c1f39a0421ab7c Reviewed-on: https://webrtc-review.googlesource.com/c/119261 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26371}
2019-01-23 12:48:13 +01:00
"../../rtc_base/experiments:rate_control_settings",
Migrate modules/video_coding to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 12:17:12 +02:00
"../../rtc_base/synchronization:mutex",
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",
In VP9 wrapper fill information required to produce Dependency Descriptor Bug: webrtc:11999 Change-Id: Id20575fca5b9279adccf1498165815aa16e044af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187340 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32421}
2020-10-15 17:26:46 +02:00
"svc:scalability_structures",
"svc:scalable_video_controller",
Delete legacy forwarding header svc_rate_allocator.h Bug: None Change-Id: I8a73f1139560b8e5a654948497751e9515aa7b92 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227029 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34581}
2021-07-27 12:48:50 +02:00
"svc:svc_rate_allocator",
Add NV12 to libvpx wrappers output Bug: webrtc:11956 Change-Id: Id8734b8f0fd87ac9b849d70b0c5764bf1ffd9c75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185300 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32225}
2020-09-29 10:37:32 +02:00
"//third_party/libyuv",
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
]
Adds field trial to set per-layer speed for libvpx vp9. The trial name WebRTC-VP9-PerLayerSpeed is used to a) set encoding speed per spatial layer, based on resolution b) allow explicitly overriding speed per layer, for testing Additionally, this CL updates the vp9 wrapper in preparation for injectable trials. Bug: webrtc:11551, webrtc:11926 Change-Id: I2bb3a664feaef60483ffc241b71070284d3e0172 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186400 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32294}
2020-10-01 19:13:13 +02:00
absl_deps = [
Support native scaling of VideoFrameBuffers in LibvpxVp9Encoder. This CL is part of Optimized Scaling efforts. In Chromium, the native frame buffer is getting an optimized CropAndScale() implementation. To support HW accelerated scaling, returning pre-scaled images and skipping unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale instead of libyuv::XXXXScale and 2) only map buffers it actually intends to encode. - To achieve this, WebRTC encoders are updated to map kNative video buffers so that in a follow-up CL VideoStreamEncoder can stop mapping intermediate buffer sizes. In this CL LibvpxVp9Encoder is updated to map kNative buffers of pixel formats it supports and convert ToI420() if the kNative buffer is something else. A fake native buffer that keeps track of which resolutions were mapped, MappableNativeBuffer, is added. Because VP9 is currently an SVC encoder and not a simulcast encoder, it does not need to invoke CropAndScale. This CL also fixes MultiplexEncoderAdapter, but because it simply forwards frames it only cares about the pixel format when |supports_augmented_data_| is true so this is the only time we map it. Because this encoder is not used with kNative in practise, we don't care to make this path optimal. Bug: webrtc:12469, chromium:1157072 Change-Id: I74edf85b18eccd0d250776bbade7a6444478efce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212580 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#33526}
2021-03-22 12:24:30 +01:00
"//third_party/abseil-cpp/absl/algorithm:container",
Adds field trial to set per-layer speed for libvpx vp9. The trial name WebRTC-VP9-PerLayerSpeed is used to a) set encoding speed per spatial layer, based on resolution b) allow explicitly overriding speed per layer, for testing Additionally, this CL updates the vp9 wrapper in preparation for injectable trials. Bug: webrtc:11551, webrtc:11926 Change-Id: I2bb3a664feaef60483ffc241b71070284d3e0172 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186400 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32294}
2020-10-01 19:13:13 +02:00
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:strings",
]
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) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("android_codec_factory_helper") {
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
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",
Move rtc_base/ignore_wundef.h to its own target. Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to move it to its own target and allow its dependant to avoid to take a dependency rtc_base:on rtc_base_approved. Bug: webrtc:9419 Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29548}
2019-10-19 12:03:23 +02:00
"../../rtc_base:ignore_wundef",
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_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) {
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
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
]
}
}
Factor encoded frame generation code into own target Show it can make vp9 tests cleaner too. Bug: None Change-Id: I8333a61dec1ef90ade9faffea94e1555ccbfcfaa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177013 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31523}
2020-06-15 12:55:22 +02:00
rtc_library("encoded_video_frame_producer") {
testonly = true
sources = [
"codecs/test/encoded_video_frame_producer.cc",
"codecs/test/encoded_video_frame_producer.h",
]
deps = [
":video_codec_interface",
"../../api:create_frame_generator",
"../../api:frame_generator_api",
"../../api/transport/rtp:dependency_descriptor",
"../../api/video:encoded_image",
"../../api/video:video_frame",
"../../api/video:video_frame_type",
"../../api/video_codecs:video_codecs_api",
"../../rtc_base:checks",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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
]
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",
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",
Move MockVideoDecoder to api/test. Move MockVideoDecoder from modules/video_coding/include/mock/mock_video_codec_interface.h to api/test/mock_video_decoder.h The mock encoder has already moved: https://webrtc-review.googlesource.com/c/src/+/105620 Keeping the old header until downstream projects have been updated. Bug: webrtc:9722 Change-Id: I4bc849173a04813064212f17761876695ca3fed4 Reviewed-on: https://webrtc-review.googlesource.com/c/105900 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25170}
2018-10-15 11:55:13 +02:00
"../../api:mock_video_decoder",
"../../api:mock_video_encoder",
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: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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_video",
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",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
]
deps = [
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
":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",
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",
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
":videocodec_test_stats_impl",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"..:module_api",
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class Bug: webrtc:10138 Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-05 15:59:00 +01:00
"../../api:create_frame_generator",
"../../api:frame_generator_api",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 14:31:24 +01:00
"../../api:sequence_checker",
Move 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",
Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da. Reason for revert: Got Aliby for FEC test flakes Original change's description: > Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" > > This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41. > > Reason for revert: Breaks downstream projects. > Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky. > > Original change's description: > > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current > > > > Bug: webrtc:10191 > > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27035} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484 > Reviewed-by: Yves Gerey <yvesg@webrtc.org> > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27041} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10191 Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 10:28:05 +00:00
"../../api/task_queue",
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 10:02:56 -08:00
"../../api/video:builtin_video_bitrate_allocator_factory",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:encoded_image",
"../../api/video:video_bitrate_allocation",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_bitrate_allocator_factory",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_video",
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 //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../rtc_base:macromagic",
Remove 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 //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../../rtc_base:rtc_event",
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",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Migrate modules/video_coding to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 12:17:12 +02:00
"../../rtc_base/synchronization:mutex",
Reland "Introduce RTC_NO_UNIQUE_ADDRESS." This is a reland of f5e261aaf65cdf2eb903cdf40d651846be44f447 This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since there have been some issues. Original change's description: > Introduce RTC_NO_UNIQUE_ADDRESS. > > This macro introduces the possibility to suggest the compiler that a > data member doesn't need an address different from other non static > data members. > > The usage of a macro is to maintain portability since at the moment > the attribute [[no_unique_address]] is only supported by clang > with at least -std=c++11 but it should be supported by all the > compilers starting from C++20. > > Bug: webrtc:11495 > Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32246} Bug: webrtc:11495, webrtc:12218 Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32668}
2020-11-23 11:07:42 +01:00
"../../rtc_base/system:no_unique_address",
Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da. Reason for revert: Got Aliby for FEC test flakes Original change's description: > Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" > > This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41. > > Reason for revert: Breaks downstream projects. > Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky. > > Original change's description: > > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current > > > > Bug: webrtc:10191 > > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27035} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484 > Reviewed-by: Yves Gerey <yvesg@webrtc.org> > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27041} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10191 Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27056}
2019-03-11 10:28:05 +00:00
"../../rtc_base/task_utils:to_queued_task",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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/ConferenceMotion_1280_720_50.yuv",
Signal requested resolution alignment requirements from sinks to sources. Bug: webrtc:11218 Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30113}
2019-12-19 09:47:11 +01:00
"../../resources/FourPeople_1280x720_30.yuv",
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
]
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
Delete header file mock_vcm_callbacks.h Move definitions of mock classes to the only user, the unit tests for the deprecated class vcm::VideoReceiver. Bug: webrtc:7408 Change-Id: I05e38ed8ebbe615bb2db0b631ec914773fb0a520 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168182 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30451}
2020-02-04 10:46:37 +01:00
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
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
}
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
}
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("videocodec_test_impl") {
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
testonly = true
sources = [
"codecs/test/videocodec_test_fixture_impl.cc",
"codecs/test/videocodec_test_fixture_impl.h",
]
deps = [
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
":codec_globals_headers",
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
":video_codec_interface",
":video_codecs_test_framework",
":video_coding_utility",
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
":videocodec_test_stats_impl",
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
":webrtc_vp9_helpers",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api:array_view",
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",
Use function_video_(en|de)coder_factory from api Remove them from test. It is completion of the move started with https://webrtc-review.googlesource.com/c/src/+/107705 Bug: None Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6 Reviewed-on: https://webrtc-review.googlesource.com/c/107891 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 15:57:48 +02:00
"../../api/test/video:function_video_factory",
Reland: Wires up WebrtcKeyValueBasedConfig in media engines. This is a reland of https://webrtc-review.googlesource.com/c/src/+/174261 Patchset 1 contains the old cl (plus a merge conflict fix). Later patchets are bufixes: A PeerConnection can be created without a Call instance (in the case of DataChannel only), so we can't always use that to fetch the current trials. Old CL descritpion: This replaces field_trial:: -based functions from system_wrappers. Field trials are still used as fallback, but injectable trials are now possible. // Since re-land is otherwise unchanged, setting previous reviewers as TBR TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org Bug: webrtc:11926 Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32163}
2020-09-22 11:36:35 +02:00
"../../api/transport:field_trial_based_config",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_bitrate_allocation",
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/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",
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 //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../rtc_base:logging",
Remove 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 //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Avoid using GlobalTaskQueueFactory for TaskQueueForTest To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor TaskQueueForTest can be used instead in tests and keep the convenient constructor. Also cleanup the TaskQueueForTest a bit: move the class to webrtc namespace add default constructor disallow copy using language construct instead of macro cleanup build dependencies rename build target (to match move out of the rtc namespace) Bug: webrtc:10284 Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27193}
2019-03-19 18:08:37 +01:00
"../../rtc_base:task_queue_for_test",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
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",
Add PrintResults to VideoCodecTest. These are used by the test runner to pick up perf values to be shown in the perf dashboard. Bug: webrtc:10349 Change-Id: Ib3b2479f7a20b66192751bee8237d757f5870bd2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126220 Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27020}
2019-03-07 14:27:57 +01:00
"../../test:perf_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
"../../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",
]
Do not crash if codec is not available Check if codec was successfully created and exit from RunTest if not before creating VideoProcessor. Bug: none Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33372}
2021-03-03 13:34:49 +01:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("videocodec_test_stats_impl") {
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
testonly = true
sources = [
"codecs/test/videocodec_test_stats_impl.cc",
"codecs/test/videocodec_test_stats_impl.h",
]
deps = [
"../../api:videocodec_test_fixture_api",
Move SampleStatsCounter to public API Bug: None Change-Id: I8956f6febbb1caf71e951d212d57746fe1ec5eb2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184506 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32142}
2020-09-18 18:23:08 +02:00
"../../api/numerics",
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
"../../rtc_base:checks",
Replace test::Statistics by webrtc::RunningStatistics. The former became redundant and didn't guarantee numerical stability for variance computation. Bug: webrtc:10412 Change-Id: Idc291abe9add41bde9e7734f179e5d6c65f2630b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132460 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27605}
2019-04-13 18:59:53 +02:00
"../../rtc_base:rtc_numerics",
Switch to RTC_LOG(LS_INFO) for non-perf VideoCodecTest text output. This allows picking up the output in Android tests, where stdout/stderr is lost but RTC_LOGs are picked up by the org.webrtc.Logging utility. Tested: Downstream Android tests. Bug: webrtc:10349 Change-Id: I1379f4303640dbc9621c64d9c88cf61bc8447ab6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132704 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27616}
2019-04-15 13:27:00 +02:00
"../../rtc_base:stringutils",
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
"../../test:test_common",
"../rtp_rtcp:rtp_rtcp_format",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("video_coding_modules_tests") {
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
testonly = true
Fix color space bug in wrapper of H264 decoder Bug: none Change-Id: I8309e5e3c177ae75712fa257e083ea2018a1f8e2 Reviewed-on: https://webrtc-review.googlesource.com/c/121760 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26597}
2019-02-07 16:42:45 +01:00
defines = []
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
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",
Reland "Add dav1d decoder to WebRTC." This reverts commit 8498b7e7f6b90fa036de2a6887d34256f0565b4f. Reason for revert: Updating CL to include conditional build flag. Original change's description: > Revert "Add dav1d decoder to WebRTC." > > This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3. > > Reason for revert: High binary size increase > > Original change's description: > > Add dav1d decoder to WebRTC. > > > > Bug: none > > Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504 > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35394} > > TBR=danilchap@webrtc.org,mbonadei@webrtc.org,ilnik@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,ssilkin@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: none > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35395} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: Iff51848731646159e87e075c38af7cb6355f5b5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238661 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35409}
2021-11-23 11:00:24 +01:00
"codecs/test/videocodec_test_av1.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",
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",
]
Reland "Add av1 test running real video clips." This reverts commit 6958d2c6f0ce5267bdc4120d88680a4be9ed5e59. Disable the test on iOS. Bug: None Change-Id: Ie42fada10a92bd4a802c6c79caeb4965410ddf6a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176461 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Jerome Jiang <jianj@google.com> Cr-Commit-Position: refs/heads/master@{#31437}
2020-06-03 20:44:50 -07:00
Fix build with rtc_libvpx_build_vp9=false Like aom and openh264, VP9 can be disabled with the gn argument. Bug: None Change-Id: I7d67e3946afae0bb4cac8a7e591445604dda9ce1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215260 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33737}
2021-04-15 07:09:58 +09:00
if (rtc_libvpx_build_vp9) {
sources += [ "codecs/vp9/test/vp9_impl_unittest.cc" ]
}
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 = [
Factor encoded frame generation code into own target Show it can make vp9 tests cleaner too. Bug: None Change-Id: I8333a61dec1ef90ade9faffea94e1555ccbfcfaa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177013 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31523}
2020-06-15 12:55:22 +02:00
":encoded_video_frame_producer",
Splits vp9_impl into libvpx_vp9_encoder and libvpx_vp9_decoder. Also moves the LibvpxVp8Interface from codec/vp8 to codec/interface and drops vp8 from the name. Follow-up CLs will wire up actual usage in the new classes through the interface so that we can unit test them more easily. Bug: webrtc:12274 Change-Id: I95f66e90245d9320e5fc23cdc845fbeb2648b38b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196522 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32816}
2020-12-09 20:58:21 +01:00
":mock_libvpx_interface",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
":video_codec_interface",
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",
Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial. Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config, and then configuring the codec in non-svc mode would cause us to not set the cpu speed in libvpx. For some reason, that could trigger a crash in the encoder. This CL fixes that, and adds new test coverage for the code affected byt the trial. Bug: chromium:1167353, webrtc:11551 Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33051}
2021-01-20 22:12:33 +01:00
":webrtc_libvpx_interface",
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",
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class Bug: webrtc:10138 Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-05 15:59:00 +01:00
"../../api:create_frame_generator",
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",
Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class Bug: webrtc:10138 Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-05 15:59:00 +01:00
"../../api:frame_generator_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",
Move MockVideoDecoder to api/test. Move MockVideoDecoder from modules/video_coding/include/mock/mock_video_codec_interface.h to api/test/mock_video_decoder.h The mock encoder has already moved: https://webrtc-review.googlesource.com/c/src/+/105620 Keeping the old header until downstream projects have been updated. Bug: webrtc:9722 Change-Id: I4bc849173a04813064212f17761876695ca3fed4 Reviewed-on: https://webrtc-review.googlesource.com/c/105900 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25170}
2018-10-15 11:55:13 +02:00
"../../api:mock_video_decoder",
"../../api:mock_video_encoder",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
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",
Use function_video_(en|de)coder_factory from api Remove them from test. It is completion of the move started with https://webrtc-review.googlesource.com/c/src/+/107705 Bug: None Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6 Reviewed-on: https://webrtc-review.googlesource.com/c/107891 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 15:57:48 +02:00
"../../api/test/video:function_video_factory",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:encoded_image",
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",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
Make rtc_software_fallback_wrappers target visible. Need to depend on them from Chromium. Bug: webrtc:7925 Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17 Reviewed-on: https://webrtc-review.googlesource.com/82061 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23580}
2018-06-12 11:15:56 +02:00
"../../api/video_codecs:rtc_software_fallback_wrappers",
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",
Refactor creation of ColorSpace test data Bug: webrtc:8651 Change-Id: I2ebb5fcdc260af19d04513ab5f3d76f81a3b4ca9 Reviewed-on: https://webrtc-review.googlesource.com/c/114282 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26012}
2018-12-14 10:14:07 +01:00
"../../common_video/test:utilities",
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",
Split out a separate target for SimulcastEncoderAdapter This will allow downstream projects to use it to construct their own injected codecs without pulling in dependencies on the software codecs. Bug: webrtc:7925 Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8 Reviewed-on: https://webrtc-review.googlesource.com/c/107027 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Jonathan Yu <yujo@chromium.org> Cr-Commit-Position: refs/heads/master@{#25297}
2018-10-19 00:51:18 -07:00
"../../media:rtc_simulcast_encoder_adapter",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../../rtc_base:refcount",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial. Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config, and then configuring the codec in non-svc mode would cause us to not set the cpu speed in libvpx. For some reason, that could trigger a crash in the encoder. This CL fixes that, and adds new test coverage for the code affected byt the trial. Bug: chromium:1167353, webrtc:11551 Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33051}
2021-01-20 22:12:33 +01:00
"../../test:explicit_key_value_config",
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",
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",
Reland "Add dav1d decoder to WebRTC." This reverts commit 8498b7e7f6b90fa036de2a6887d34256f0565b4f. Reason for revert: Updating CL to include conditional build flag. Original change's description: > Revert "Add dav1d decoder to WebRTC." > > This reverts commit 147858577d4db6d257d3cc248fe571a1bbf887e3. > > Reason for revert: High binary size increase > > Original change's description: > > Add dav1d decoder to WebRTC. > > > > Bug: none > > Change-Id: I7642f42e592dcf510679f881f118bc4dab93b31c > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237504 > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Magnus Flodman <mflodman@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35394} > > TBR=danilchap@webrtc.org,mbonadei@webrtc.org,ilnik@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,ssilkin@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I00a8acd6ea94ce523c2d5ba705333c9174678180 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: none > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238560 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35395} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: Iff51848731646159e87e075c38af7cb6355f5b5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238661 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35409}
2021-11-23 11:00:24 +01:00
"codecs/av1:dav1d_decoder",
"codecs/av1:libaom_av1_decoder",
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
]
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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
}
Don't increment timestamp on drop/reencode in LibvpxVp8Encoder. I don't think this has any impact, just wanted to have a first unit test to play around with. Bug: None Change-Id: I892e2642f0243c5c9ee807cf71abcd96112b25f4 Reviewed-on: https://webrtc-review.googlesource.com/c/105000 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25089}
2018-10-10 14:58:46 +02:00
if (rtc_build_libvpx) {
deps += [ rtc_libvpx_dir ]
}
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 source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("video_coding_unittests") {
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
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 = [
Introduce ChainDiffCalculator to convert flags which chains a video frame part of into chain_diffs Bug: webrtc:10342 Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31306}
2020-05-18 14:14:11 +02:00
"chain_diff_calculator_unittest.cc",
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",
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",
New FrameBuffer3. FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe. Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35494}
2021-12-07 18:13:09 +01:00
"frame_buffer3_unittest.cc",
Add helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
"frame_dependencies_calculator_unittest.cc",
Reland "Copy video frames metadata between encoded and plain frames in one place" Reland with fixes. Currently some video frames metadata like rotation or ntp timestamps are copied in every encoder and decoder separately. This CL makes copying to happen at a single place for send or receive side. This will make it easier to add new metadata in the future. Also, added some missing tests. Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 Bug: webrtc:10460 Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27930}
2019-05-13 16:13:36 +02:00
"generic_decoder_unittest.cc",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
"h264_packet_buffer_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",
Migrate VCMInterFrameDelay to use Time units Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-07 14:50:51 +01:00
"inter_frame_delay_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
"jitter_buffer_unittest.cc",
"jitter_estimator_tests.cc",
Create LossNotificationController Create LossNotificationController, which produces LossNotification RTCP feedback messages when video packets/frames are lost. (LossNotification messages are sent when an RTP gap is detected, as well as when frames are later received which are undecodable because of the missing frames due to the previously dropped packets.) Bug: webrtc:10336 Change-Id: I7b3a156ed14e5a727349acdd82dae6997462421b Reviewed-on: https://webrtc-review.googlesource.com/c/123762 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26812}
2019-02-21 16:25:40 +01:00
"loss_notification_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
"nack_module_unittest.cc",
NackModule2: Rename to NackRequester. The alternative new name proposed, NackTracker, is already in use in audio_coding. Fixed: webrtc:11594 Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-22 15:09:39 +02:00
"nack_requester_unittest.cc",
Rename PacketBuffer tests to follow conventions Rename file with tests to match code under test. Rename fixture by moving 'Test' from prefix to suffix Bug: None Change-Id: I54c36d3b517bde7cdffa3a7e74528cc464ea7ad7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157301 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29532}
2019-10-17 19:06:46 +02:00
"packet_buffer_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
"receiver_unittest.cc",
"rtp_frame_reference_finder_unittest.cc",
VP8 ref finder unittest cleanup Change-Id: I627dda1229ceb4b2da3f37f7418da7b7653e4d04 Bug: webrtc:12221 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208482 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33362}
2021-03-01 17:15:03 +01:00
"rtp_vp8_ref_finder_unittest.cc",
Move RtpVp9RefFinder unittests into its own file. Bug: webrtc:12221 Change-Id: I72bb25503d2894aae596426cb40764246c087767 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196342 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32760}
2020-12-03 14:47:23 +01:00
"rtp_vp9_ref_finder_unittest.cc",
Switch VCMRttFilter to use TimeDelta * Moved into its own GN target * Switched the internal buffer types to absl::InlinedVector as arrays are tricky to use with types that do not have default constructors. * Update fields arnd variables to use style guide. * Use constexpr for formerly const fields. * Adds unit tests. Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 15:22:08 +01:00
"rtt_filter_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
"session_info_unittest.cc",
"test/stream_generator.cc",
"test/stream_generator.h",
Fix dropped frames not counted issue There's been reports of dropped frames that are not counted and correctly reported by getStats(). If a HW decoder is used and the system is provoked by stressing the system, I've been able to reproduce this problem. It turns out that we've missed frames that are dropped because there is no callback to the Decoded() function. This CL restructures the code so that dropped frames are counted even in cases where there's no corresponding callback for some frames. Bug: webrtc:11229 Change-Id: I0216edba3733399c188649908d459ee86a9093d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214783 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33671}
2021-04-09 16:03:22 +02:00
"timestamp_map_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
"timing_unittest.cc",
Split out counting unique rtp timestamps from packet_buffer Bug: None Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29656}
2019-10-30 14:12:24 +01:00
"unique_timestamp_counter_unittest.cc",
video: Implement bandwidth based scaler The |slice_qp_detla| reported by the hardware is not credible, which causing the quality scaler cannot work properly,the resolution cannot be adjusted correctly. To fix this issue, this CL implements a bandwidth scaler which is used for adjust resolution, this scaler will be used when QP based quality scaler is not working due to untrusted QP reported by HW AVC encoder. Bug: webrtc:12942 Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 17:19:44 +08:00
"utility/bandwidth_quality_scaler_unittest.cc",
Introduce DecodedFramesHistory class and use it in FrameBuffer This is a space efficient way to store more records about decoded frames, which is needed for long term references. Bug: webrtc:9710 Change-Id: I051d59d34a966d48db011142466d9cd15304b7d9 Reviewed-on: https://webrtc-review.googlesource.com/c/116792 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26240}
2019-01-14 13:24:22 +01:00
"utility/decoded_frames_history_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/frame_dropper_unittest.cc",
Move frame drop functionality in VideoAdapter into a separate class. This class will replace modules/video_coding/utility/framerate_controller.h Bug: webrtc:13031 Change-Id: I8faa9c3c158b7c5ab0618e3504224c7e00f8e0b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227350 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34802}
2021-08-16 09:33:13 +02:00
"utility/framerate_controller_deprecated_unittest.cc",
Introduce IVF file reader Bug: webrtc:10138 Change-Id: I97d332942f4e645527330159efefb1cb1d8034a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160008 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29844}
2019-11-20 13:30:19 +01:00
"utility/ivf_file_reader_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",
Reland "Parse encoded frame QP if not provided by encoder" This reverts commit 727d2afc4330efebc904e0e4f366e885d7b08787. Reason for revert: Use thread-safe wrapper for H264 parser. Original change's description: > Revert "Parse encoded frame QP if not provided by encoder" > > This reverts commit 8639673f0c098efc294a7593fa3bd98e28ab7508. > > Reason for revert: linux_tsan fails https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25329/overview > > Original change's description: > > Parse encoded frame QP if not provided by encoder > > > > Bug: webrtc:12542 > > Change-Id: Ic70b46e226f158db7a478a9f20e1f940804febba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210966 > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33434} > > TBR=asapersson@webrtc.org,ssilkin@webrtc.org > > Change-Id: Ie251d8f70f8e87fd86b63730aefd2ef3f941e4bb > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12542 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211355 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33441} # Not skipping CQ checks because this is a reland. Bug: webrtc:12542 Change-Id: Ib7601fd6f2f26bceddbea2b4ba54d67a281f3a59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211660 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33458}
2021-03-15 10:12:57 +01:00
"utility/qp_parser_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/quality_scaler_unittest.cc",
"utility/simulcast_rate_allocator_unittest.cc",
VP9 parser: undo r34393 and fix incorrect return statement. Some code was deleted in https://webrtc-review.googlesource.com/c/src/+/224266/2/modules/video_coding/utility/vp9_uncompressed_header_parser.cc since it was detected as unreachable. The root cause was an early return that should have been a RETURN_IF_FALSE(x). Bug: webrtc:12924 Change-Id: Ifadded9bbb4748d56cf65c30fd8f87e92fde10d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225040 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34422}
2021-07-06 15:05:24 +02:00
"utility/vp9_uncompressed_header_parser_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_receiver_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 = [
Introduce ChainDiffCalculator to convert flags which chains a video frame part of into chain_diffs Bug: webrtc:10342 Change-Id: I6fb899eae934078223b101c9f85e2ac101980d4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175108 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31306}
2020-05-18 14:14:11 +02:00
":chain_diff_calculator",
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",
New FrameBuffer3. FrameBuffer3 keep track of order, decodability and continuity of the inserted frames. Compared to FrameBuffer2 which schedule frames for decoding and is thread safe, FrameBuffer3 does not schedule decoding and is thread unsafe. Change-Id: Ic3bd540c4f69cec26fce53a40425f3bcd9afe085 Bug: webrtc:13343 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238985 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35494}
2021-12-07 18:13:09 +01:00
":frame_buffer",
Add helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
":frame_dependencies_calculator",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
":h264_packet_buffer",
Migrate VCMInterFrameDelay to use Time units Additionally, * Moved to its own GN target. * Added unittests. * Removed unused variable `_zeroWallClock`. * Renamed variables to match style guide. * Moved fields _dTS and _wrapArounds to variables. Change-Id: I7aa8b8dec55abab49ceabe838dabf2a7e13d685d Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253580 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36147}
2022-03-07 14:50:51 +01:00
":inter_frame_delay",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
":jitter_estimator",
NackModule2: Rename to NackRequester. The alternative new name proposed, NackTracker, is already in use in audio_coding. Fixed: webrtc:11594 Change-Id: I6a05fafc05fa7ddb18ea4f64886a135e5ef59f7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226744 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34539}
2021-07-22 15:09:39 +02:00
":nack_requester",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
":packet_buffer",
Switch VCMRttFilter to use TimeDelta * Moved into its own GN target * Switched the internal buffer types to absl::InlinedVector as arrays are tricky to use with types that do not have default constructors. * Update fields arnd variables to use style guide. * Use constexpr for formerly const fields. * Adds unit tests. Change-Id: I476ae8491f0f9878c176e7b87a5133942c3d79f7 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253120 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36133}
2022-03-04 15:22:08 +01:00
":rtt_filter",
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",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
":timing",
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",
Move code related to VideoCodingModule to its own build target The new target, modules/video_coding:video_coding_legacy, is not depended upon by any webrtc non-test code. Bug: webrtc:7408 Change-Id: I94127e2b8b3b8f15917bfa38e602f8face91fcdb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152163 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29133}
2019-09-10 10:48:48 +02:00
":video_coding_legacy",
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_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",
Change VideoCodecTestStats API. - Add GetFrameStatistics API: This is useful for downstream test users that want to read frame-level stats. - Remove other APIs that are not used by downstream tests: * AddFrame * GetFrame * GetFrameWithTimestamp * SliceAndCalcAggregatedVideoStatistic * PrintFrameStatistics * Size * Clear The implementations, which are used by the fixture implementation, are kept. Bug: webrtc:10349 Change-Id: Id2f6fa5a36b8341a5ccb365725f71ebe0c0f1570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128779 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27306}
2019-03-26 16:56:14 +01:00
":videocodec_test_stats_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",
Move TemporalLayers to api/video_codecs. Also renaming it Vp8TemporalLayers to show that it is codec specific. Bug: webrtc:9012 Change-Id: I18187538b8142cdd7538f1a4ed1bada09d040f1f Reviewed-on: https://webrtc-review.googlesource.com/c/104643 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25137}
2018-10-12 10:30:31 +02:00
":webrtc_vp8_temporal_layers",
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_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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"..:module_fec_api",
"../../api:array_view",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api:fec_controller_api",
Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create Previously, FecControllerOverride was passed to Vp8FrameBufferController::SetFecControllerOverride. Passing to the factory is a more elegant way, since it's only used when the controller is constructed. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28443}
2019-07-02 11:20:09 +02:00
"../../api:mock_fec_controller_override",
Move MockVideoDecoder to api/test. Move MockVideoDecoder from modules/video_coding/include/mock/mock_video_codec_interface.h to api/test/mock_video_decoder.h The mock encoder has already moved: https://webrtc-review.googlesource.com/c/src/+/105620 Keeping the old header until downstream projects have been updated. Bug: webrtc:9722 Change-Id: I4bc849173a04813064212f17761876695ca3fed4 Reviewed-on: https://webrtc-review.googlesource.com/c/105900 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25170}
2018-10-15 11:55:13 +02:00
"../../api:mock_video_decoder",
"../../api:mock_video_encoder",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
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: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",
Reland "Copy video frames metadata between encoded and plain frames in one place" Reland with fixes. Currently some video frames metadata like rotation or ntp timestamps are copied in every encoder and decoder separately. This CL makes copying to happen at a single place for send or receive side. This will make it easier to add new metadata in the future. Also, added some missing tests. Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 Bug: webrtc:10460 Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27930}
2019-05-13 16:13:36 +02:00
"../../api/task_queue:default_task_queue_factory",
Use function_video_(en|de)coder_factory from api Remove them from test. It is completion of the move started with https://webrtc-review.googlesource.com/c/src/+/107705 Bug: None Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6 Reviewed-on: https://webrtc-review.googlesource.com/c/107891 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25392}
2018-10-26 15:57:48 +02:00
"../../api/test/video:function_video_factory",
Use TimeDelta and Timestamp in VCMJitterEstimator * Uses DataSize to represent incoming and outgoing bytes. * Puts units into doubles as they enter the Kalman filter * Moved to its own GN target. Change-Id: I1e7d5486a00a7158d418f553a6c77f9dd56bf3c2 Bug: webrtc:13756 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253121 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36143}
2022-03-07 13:21:51 +01:00
"../../api/units:data_size",
Use Timestamp and TimeDelta in VCMTiming * Switches TimestampExtrapolator to use Timestamp as well. Bug: webrtc:13589 Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:13:55 +01:00
"../../api/units:frequency",
Implement FrameBuffer3Proxy This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
2021-12-12 15:27:00 +01:00
"../../api/units:time_delta",
Use Timestamp and TimeDelta in VCMTiming * Switches TimestampExtrapolator to use Timestamp as well. Bug: webrtc:13589 Change-Id: I042be5d693068553d2e8eb92fa532092d77bd7ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249993 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36112}
2022-03-02 15:13:55 +01:00
"../../api/units:timestamp",
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 10:02:56 -08:00
"../../api/video:builtin_video_bitrate_allocator_factory",
VP8 ref finder unittest cleanup Change-Id: I627dda1229ceb4b2da3f37f7418da7b7653e4d04 Bug: webrtc:12221 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208482 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33362}
2021-03-01 17:15:03 +01:00
"../../api/video:encoded_frame",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
"../../api/video:render_resolution",
Unify AdaptationReason and AdaptReason enums. Moves the unified AdaptationReason to the api/ folder. Bug: webrtc:11392 Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583 Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 11:36:55 +02:00
"../../api/video:video_adaptation",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/video:video_bitrate_allocation",
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 "Use the factory instead of using the builtin code path in `VideoCodecInitializer`" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 10:02:56 -08:00
"../../api/video:video_bitrate_allocator_factory",
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 helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
"../../api/video:video_frame_type",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Use single FrameBufferController in VP8, created by a factory. This CL paves the way to making FrameBufferController injectable. LibvpxVp8Encoder can manage multiple streams. Prior to this CL, each stream had its own frame buffer controller, all of them held in a vector by LibvpxVp8Encoder. This complicated the code and produced some code duplication (cf. SetupTemporalLayers). This CL: 1. Replaces CreateVp8TemporalLayers() by a factory. (Later CLs will make this factory injectable.) 2. Makes LibvpxVp8Encoder use a single controller. This single controller will, in the case of multiple streams, delegate its work to multiple controllers, but that fact is not visible to LibvpxVp8Encoder. This CL also squashes CL #126046 (Send notifications of RTT and PLR changes to Vp8FrameBufferController) into it. Bug: webrtc:10382 Change-Id: Id9b55734bebb457acc276f34a7a9e52cc19c8eb9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126483 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27206}
2019-03-20 11:56:20 +01:00
"../../api/video_codecs:vp8_temporal_layers_factory",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../common_video",
Add helper to calculate frame dependencies based on encoder buffer usage Bug: webrtc:10342 Change-Id: I1d856d060c2defcd10310f0d8639ce8a9554fff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168194 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30458}
2020-02-05 15:06:17 +01:00
"../../common_video/generic_frame_descriptor",
Reland "Copy video frames metadata between encoded and plain frames in one place" Reland with fixes. Currently some video frames metadata like rotation or ntp timestamps are copied in every encoder and decoder separately. This CL makes copying to happen at a single place for send or receive side. This will make it easier to add new metadata in the future. Also, added some missing tests. Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 Bug: webrtc:10460 Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27930}
2019-05-13 16:13:36 +02:00
"../../common_video/test:utilities",
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",
Cleanup BUILD.gn files from imports like foo:foo Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn files. Done with Sublime regex replace. Find: \b([-a-zA-Z0-9_]+):+\1\b In: *.gn Replace with: \1 Bug: None Change-Id: I40aba1b14face687a595b852ffe443cb20197611 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:35:10 +01:00
"../../rtc_base",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove //rtc_base:refcount from public deps Bug: webrtc:8603 Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 02:54:12 +02:00
"../../rtc_base:refcount",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
Reland 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",
Remove //rtc_base:rtc_event from public deps Bug: webrtc:8603 Change-Id: Ib99f43043da17723c939b0fe2aa9f3e515462c93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257911 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36450}
2022-04-04 16:57:52 +02:00
"../../rtc_base:rtc_event",
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",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../rtc_base:stringutils",
Avoid using GlobalTaskQueueFactory for TaskQueueForTest To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor TaskQueueForTest can be used instead in tests and keep the convenient constructor. Also cleanup the TaskQueueForTest a bit: move the class to webrtc namespace add default constructor disallow copy using language construct instead of macro cleanup build dependencies rename build target (to match move out of the rtc namespace) Bug: webrtc:10284 Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27193}
2019-03-19 18:08:37 +01:00
"../../rtc_base:task_queue_for_test",
Remove //rtc_base:timeutils from public deps Bug: webrtc:8603 Change-Id: Iaca9356d16275a02e8842c783f259131d72ef010 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257914 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36460}
2022-04-04 17:16:15 +02:00
"../../rtc_base:timeutils",
video: Implement bandwidth based scaler The |slice_qp_detla| reported by the hardware is not credible, which causing the quality scaler cannot work properly,the resolution cannot be adjusted correctly. To fix this issue, this CL implements a bandwidth scaler which is used for adjust resolution, this scaler will be used when QP based quality scaler is not working due to untrusted QP reported by HW AVC encoder. Bug: webrtc:12942 Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 17:19:44 +08:00
"../../rtc_base/experiments:encoder_info_settings",
JitterEstimator: Remove old LowRate exp and add trial for upper bound. The frame time deltas are now capped based on the current noise. This has been tested in various conditions using both screen content and typical mobile video settings, to produce delays that are not overly high screen content, and simultaneously not negatively affect mobile calls on really bad network that may have high natural jitter. Bug: webrtc:9898 Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48 Reviewed-on: https://webrtc-review.googlesource.com/c/107349 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25469}
2018-11-01 11:20:49 +01:00
"../../rtc_base/experiments:jitter_upper_bound_experiment",
Migrate modules/video_coding to webrtc::Mutex. Bug: webrtc:11567 Change-Id: I8023fbe7595f7ba8ae7c7db3583fc2e560ec3df2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178803 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31644}
2020-07-07 12:17:12 +02:00
"../../rtc_base/synchronization:mutex",
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class. Bug: webrtc:12579 Change-Id: Idea35983e204e4a3f8628d5b4eb587bbdbff5877 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227286 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34999}
2021-09-15 10:45:43 +02:00
"../../rtc_base/system:unused",
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",
Reland "Copy video frames metadata between encoded and plain frames in one place" Reland with fixes. Currently some video frames metadata like rotation or ntp timestamps are copied in every encoder and decoder separately. This CL makes copying to happen at a single place for send or receive side. This will make it easier to add new metadata in the future. Also, added some missing tests. Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346 Bug: webrtc:10460 Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27930}
2019-05-13 16:13:36 +02:00
"../../test:fake_video_codecs",
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",
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 10/inf This patch takes a stab at modules/video_coding, but reaches only about half. Bug: webrtc:10335 Change-Id: I0d47d0468b818145470c51ae4e8e75ff58d499ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256112 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36335}
2022-03-25 12:43:14 +01:00
"../../test:scoped_key_value_config",
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",
Remove unused NextFrame function from FrameBuffer. Also updated FrameBuffer unittests to use the GlobalSimulatedTimeController. Bug: webrtc:7408, webrtc:9378 Change-Id: I8ade27492f66cdd8950b38f5f4a268714dbc35fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164536 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30422}
2020-01-29 17:36:11 +01:00
"../../test/time_controller:time_controller",
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",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../rtp_rtcp:rtp_video_header",
Restore setting encoder speed for AV1 encoder wrapper Also add simple unittests for the wrapper. Bug: webrtc:11404 Change-Id: I41d185da9bce392297d1982194c059bddb7881ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171481 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30867}
2020-03-24 09:35:04 +01:00
"codecs/av1:video_coding_codecs_av1_tests",
Deprecate webrtc::NackModule. This CL moves webrtc::NackModule to a deprecated folder and annotates the type with RTC_DEPRECATED. Since the header should not be used outside of WebRTC, this CL doesn't created a forward header. Bug: webrtc:11611 Change-Id: I4d5899d473d78b8c7f4a6a018e2805648244b5f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176360 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31394}
2020-05-30 17:31:46 +02:00
"deprecated:nack_module",
Move scalability structures from av1 into own subfolder To make it natural to reuse them for vp9 Bug: webrtc:11999 Change-Id: If2ef7ca16b8be96e0e03bb19211d9f5eb74b2d3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188620 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32414}
2020-10-15 15:57:17 +02:00
"svc:scalability_structure_tests",
Move SvcRateAllocator out of codecs/vp9 and make it not vp9 specific to reuse it for scalable av1 Bug: webrtc:12148 Change-Id: I0d413c5466a2d66df3eea9cdf13687f4ddd1a843 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191765 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32560}
2020-11-05 18:06:00 +01:00
"svc:svc_rate_allocator_tests",
Use absl_deps in order to preapre to the Abseil component build release. Bug: webrtc:1046390 Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-05 14:30:41 +02:00
]
absl_deps = [
Use 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",
JitterEstimator: Remove old LowRate exp and add trial for upper bound. The frame time deltas are now capped based on the current noise. This has been tested in various conditions using both screen content and typical mobile video settings, to produce delays that are not overly high screen content, and simultaneously not negatively affect mobile calls on really bad network that may have high natural jitter. Bug: webrtc:9898 Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48 Reviewed-on: https://webrtc-review.googlesource.com/c/107349 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25469}
2018-11-01 11:20:49 +01:00
"//third_party/abseil-cpp/absl/types:optional",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"//third_party/abseil-cpp/absl/types:variant",
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 ]
}
}
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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