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webrtc_m130/modules/video_coding/BUILD.gn

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GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../../webrtc.gni")
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
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",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
"../../modules:module_api",
"../../modules:module_api_public",
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",
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",
"//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",
"../../api/video:video_frame_type",
"../../common_video/generic_frame_descriptor",
"../../rtc_base:checks",
"../../rtc_base:logging",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("nack_module") {
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",
Fork NackModule and RtpVideoStreamReceiver Bug: webrtc:11595 Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31337}
2020-05-22 17:36:36 +02:00
"nack_module2.cc",
"nack_module2.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",
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",
"../../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/synchronization:sequence_checker",
"../../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",
Adds a field trial parameter to configure waiting time before sending Nack packets. Currently we send Nack as soon as we see packets out of order(a skip in packet sequence number). Sometimes this is not necessary because these "missing" packets just late for a couple of millisecond, or these packets can be recovered by FEC. This CL add a field trial parameter to configure a delay before sending Nack. Bug: webrtc:9953 Change-Id: Ia8f5995d874f7c55a74091bc90d8395b9b88e66b Reviewed-on: https://webrtc-review.googlesource.com/c/109080 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25488}
2018-11-02 14:51:15 +01:00
"../../system_wrappers:field_trial",
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
]
}
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") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
deps = [
[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",
Detach H264 sps pps tracker from VCMPacket Bug: webrtc:10979 Change-Id: I6ec5db570c3957dd068109accad88d2f62304163 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158523 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@{#29639}
2019-10-28 13:27:05 +01:00
"../../api:array_view",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
[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",
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",
"../../api/video:video_bitrate_allocator_factory",
Do not propagate generic descriptor on receiving frame It was used only for the frame decryptor. Decryptor needs only raw representation that it can recreate in a way compatible with the new version of the descriptor. This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f. with adjustments. Change-Id: I935977179bef31d8e1023964b967658e9a7db92d Bug: webrtc:10342 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30532}
2020-02-12 19:25:57 +01:00
"../../rtc_base:deprecation",
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",
"../../system_wrappers:metrics",
[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",
change PacketBuffer to return it's result rather that use callback Bug: None Change-Id: I8cc05dd46e811d6db37af520d2106af21c671def Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157893 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@{#29589}
2019-10-22 17:12:42 +02:00
"//third_party/abseil-cpp/absl/base:core_headers",
[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/memory",
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
]
Enable building WebRTC without built-in software codecs This CL adds a GN build flag to include builtin software codecs (enabled by default). When setting the flag to false, libvpx can also be excluded. The benefit is that the resulting binary is smaller. Replaces https://webrtc-review.googlesource.com/c/src/+/29203 Bug: webrtc:7925 Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10 Reviewed-on: https://webrtc-review.googlesource.com/36340 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-30 10:32:13 +01:00
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
sources = [
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"codec_timer.cc",
"codec_timer.h",
Split VCMCodecDataBase into VCMEncoderDataBase and VCMDecoderDataBase. Intended to ease further refactoring, cleanup and deletion in this code. Bug: webrtc:8830 Change-Id: Ib862b073e93b67b4f8eedbbf40ad3a8354a566a2 Reviewed-on: https://webrtc-review.googlesource.com/55562 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22120}
2018-02-20 16:09:48 +01:00
"decoder_database.cc",
"decoder_database.h",
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
"inter_frame_delay.cc",
"inter_frame_delay.h",
"internal_defines.h",
"jitter_estimator.cc",
"jitter_estimator.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",
Packet buffer for the new jitter buffer. BUG=webrtc:5514 R=stefan@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1772383002 Cr-Commit-Position: refs/heads/master@{#12194}
2016-04-01 02:01:54 -07:00
"packet_buffer.cc",
"packet_buffer.h",
Logic for finding frame references moved from PacketBuffer to new class RtpFrameReferenceFinder. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/1961053002 Cr-Commit-Position: refs/heads/master@{#12725}
2016-05-13 06:01:03 -07:00
"rtp_frame_reference_finder.cc",
"rtp_frame_reference_finder.h",
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
2015-11-18 22:00:21 +01:00
"rtt_filter.cc",
"rtt_filter.h",
"timestamp_map.cc",
"timestamp_map.h",
"timing.cc",
"timing.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
]
Enable building WebRTC without built-in software codecs This CL adds a GN build flag to include builtin software codecs (enabled by default). When setting the flag to false, libvpx can also be excluded. The benefit is that the resulting binary is smaller. Replaces https://webrtc-review.googlesource.com/c/src/+/29203 Bug: webrtc:7925 Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10 Reviewed-on: https://webrtc-review.googlesource.com/36340 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-30 10:32:13 +01:00
deps += [
Move SPS/PPS/IDR requirement from RtpFrameObject to PacketBuffer. BUG=webrtc:8423 Change-Id: I0f0d59461afead700c20c9a2ed9b2bc991590b4a Reviewed-on: https://webrtc-review.googlesource.com/15101 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20559}
2017-11-02 14:28:06 +01:00
":codec_globals_headers",
Create public EncodedFrame interface. The plan is to: 1. Move FrameObject to api/video. 2. Rename FrameObject to EncodedFrame. 3. Move EncodedFrame out of the video_coding namespace. This is the 1st CL. Bug: webrtc:8909 Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4 Reviewed-on: https://webrtc-review.googlesource.com/55560 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 14:30:34 +01:00
":encoded_frame",
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",
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",
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",
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",
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",
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",
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",
Exposing video bitrate allocator into API In order to have public video bitrate allocator factory, the video bitrate allocator has be part of the api. Bug: webrtc:9513 Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2 Reviewed-on: https://webrtc-review.googlesource.com/88083 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-20 15:49:43 -07:00
"../../api/video:video_bitrate_allocator",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
"../../api/video:video_frame_i420",
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 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",
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",
Replacing SequencedTaskChecker with SequenceChecker. Bug: webrtc:9883 Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 13:44:04 +02:00
"../../rtc_base/synchronization:sequence_checker",
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",
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",
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",
Make FrameBuffer support an unlimited number of dependents per frame Bug: webrtc:10190 Change-Id: I59680ec0dc05bc77dcbef50ddbb83ce2bcd91f7e Reviewed-on: https://webrtc-review.googlesource.com/c/116788 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26196}
2019-01-10 15:02:54 +01:00
"//third_party/abseil-cpp/absl/container:inlined_vector",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. Bug: none Change-Id: If28f57c5ae250afbb47c5d20c9854e9a11182642 Reviewed-on: https://webrtc-review.googlesource.com/87561 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23904}
2018-07-09 15:52:29 +02:00
"//third_party/abseil-cpp/absl/types:variant",
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
]
}
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",
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
"..:module_api",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video: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",
[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",
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",
":video_codec_interface",
":video_coding",
"..:module_api",
"..:module_api_public",
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
"../../:webrtc_common",
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",
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",
"../../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",
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/synchronization:sequence_checker",
"../../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",
"//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 = [
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 VP9 frame rate controller to separate class. Bug: webrtc:9669 Change-Id: I6f30587778e9783182af11d2410464024918e171 Reviewed-on: https://webrtc-review.googlesource.com/96201 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24487}
2018-08-29 01:05:26 +02:00
"utility/framerate_controller.cc",
"utility/framerate_controller.h",
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",
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
"utility/quality_scaler.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/quality_scaler.h",
Generalize SimulcastEncoderAdapter, use for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 16:16:38 +02:00
"utility/simulcast_rate_allocator.cc",
"utility/simulcast_rate_allocator.h",
"utility/simulcast_utility.cc",
"utility/simulcast_utility.h",
Hardware VP8 encoding: Use QP as metric for resize. Add vp8 frame header parser to get QP from vp8 bitstream. BUG= 4273 R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49259004 Cr-Commit-Position: refs/heads/master@{#9256}
2015-05-21 11:12:02 -07:00
"utility/vp8_header_parser.cc",
modules/video_coding/utility: Remove include This makes it clearer this code not meant to be used as an API. I could not find any use of this in downstream code. BUG=webrtc:5095 TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org TBR=magjed@webrtc.org Review URL: https://codereview.webrtc.org/1440873005 . Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 23:04:10 +01:00
"utility/vp8_header_parser.h",
Add vp9 QP parser. BUG=webrtc:7662 Review-Url: https://codereview.webrtc.org/2891803003 Cr-Commit-Position: refs/heads/master@{#18260}
2017-05-24 10:00:16 -07:00
"utility/vp9_uncompressed_header_parser.cc",
"utility/vp9_uncompressed_header_parser.h",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps = [
Add untracked headers to video_coding. This creates a new target for pure defines and interfaces. I think that makes sense (though include/ makes it harder to see when .cc and .h files should live together). Bug: webrtc:7620 Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e Reviewed-on: https://webrtc-review.googlesource.com/34657 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21516}
2018-01-08 11:05:10 +01:00
":video_codec_interface",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
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",
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 of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) Reason for revert: Reland with temporary deprecated API to not break chromium and google3. Original issue's description: > Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) > > Reason for revert: > Suspect of breaking Chrome FYI bots. > > See > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065 > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder > > Example logs: > ../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory > #include "third_party/webrtc/video_encoder.h" > ^ > > Original issue's description: > > Move video_encoder.h and video_decoder.h to /api and create GN targets for them > > > > BUG=webrtc:5881 > > # Because PRESUBMIT ignores LINT blacklist for moved files and these > > # headers have some not easy to resolve issues. > > NOPRESUBMIT=True > > > > Review-Url: https://codereview.webrtc.org/2780943003 > > Cr-Commit-Position: refs/heads/master@{#17511} > > Committed: https://chromium.googlesource.com/external/webrtc/+/c42f54057050c933008a49d57582577bfb9aed25 > > TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5881 > > Review-Url: https://codereview.webrtc.org/2794033002 > Cr-Commit-Position: refs/heads/master@{#17514} > Committed: https://chromium.googlesource.com/external/webrtc/+/716d7ac5c1ed6e392e264b34065800bbf03772b3 TBR=solenberg@webrtc.org,sprang@webrtc.org,guidou@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5881 Review-Url: https://codereview.webrtc.org/2795163002 Cr-Commit-Position: refs/heads/master@{#17537}
2017-04-05 03:02:20 -07:00
"../../api/video_codecs:video_codecs_api",
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed. (patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ )" This reverts commit efd902cb1d9bbd81247a3e168f2080beae761d78. Originally reviewed in https://codereview.webrtc.org/2149553002 The uptream problem should be fixed by https://codereview.webrtc.org/2145393003/ BUG=webrtc:5687 TBR=tommi@webrtc.org Review-Url: https://codereview.webrtc.org/2152013002 Cr-Commit-Position: refs/heads/master@{#13483}
2016-07-14 23:35:55 -07:00
"../../common_video",
Stop using public_deps in modules/rtp_rtcp. TBR=phoglund@webrtc.org Bug: webrtc:8603 Change-Id: I86830df23db3f33a1a26098e639596bd3b86485a No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/29780 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21108}
2017-12-06 07:51:33 +01:00
"../../modules/rtp_rtcp",
Fix circular dependencies between optional, array_view, and rtc_base. This splits things out of rtc_base and makes dependencies explicit. Bug: webrtc:6828 Change-Id: Id521896c3c43595349021c857bec216e429a0c8d Reviewed-on: https://webrtc-review.googlesource.com/32780 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21264}
2017-12-13 16:05:42 +01:00
"../../rtc_base:checks",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../rtc_base:rtc_task_queue",
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",
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",
Replacing SequencedTaskChecker with SequenceChecker. Bug: webrtc:9883 Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 13:44:04 +02:00
"../../rtc_base/synchronization:sequence_checker",
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",
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",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
]
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
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",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../../media:rtc_h264_profile_id",
Enable include check for webrtc_h264 target BUG=webrtc:7242,webrtc:6828 Review-Url: https://codereview.webrtc.org/2718143002 Cr-Commit-Position: refs/heads/master@{#16888}
2017-02-28 00:28:04 -08:00
"../../media:rtc_media_base",
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",
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",
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",
Generalize SimulcastEncoderAdapter, use for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 16:16:38 +02:00
"//third_party/libyuv",
iOS HW H264 support. First step towards supporting H264 on iOS. More tuning/experimentation required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini. Future work to get it working on OS/X, simulator (renders black screen currently) and with the Android AppRTCDemo. Currently protected with a compile time guard. BUG=4081 R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1187573004. Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 14:34:58 -07:00
]
H.264: Default flags and pulling in openh264 and ffmpeg. Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg). These will be used in a follow-up CL that adds an encoder and decoder implementation. The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added. BUG=468365 Review URL: https://codereview.webrtc.org/1575913003 Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 10:19:02 -08:00
rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni. BUG=468365 NOTRY=True Review URL: https://codereview.webrtc.org/1601813005 Cr-Commit-Position: refs/heads/master@{#11333}
2016-01-21 03:34:40 -08:00
if (rtc_use_h264) {
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",
"..:module_api",
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:19:43 +02:00
"../../api:fec_controller_api",
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:26:48 +01:00
"../../api:scoped_refptr",
Move EncodedImage class to api/video/ Bug: webrtc:9378 Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba Reviewed-on: https://webrtc-review.googlesource.com/c/104002 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Peter Slatala <psla@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-05 14:17:58 +02:00
"../../api/video:encoded_image",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame",
"../../api/video:video_frame_i420",
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",
Reland "Wrap Alpha and YUV frame into one EncodedImage for transmission" This reverts commit d756fd06fed1b6c65dcb263cbd8f00ca23d72f3b. Original change's description: > Revert "Wrap Alpha and YUV frame into one EncodedImage for transmission" > > This reverts commit 5670c86aeccc9bc1191725431de7998d21b73c07. > > Reason for revert: Breaks downstream build. Need to add "#include <cstring>" to stereo_encoder_adapter.cc to use std::memcpy. > > Original change's description: > > Wrap Alpha and YUV frame into one EncodedImage for transmission > > > > With alpha channel, we observe the artifacts on the receiver side, and > > the reason is that when YUV channel has a key frame, it gives frame_buffer2 > > a chance to drop some previous frames. Then it is possible that some alpha > > frames got dropped, which break the alpha frame dependence chain. > > > > In this CL, we pack the YUV frame and alpha encoded frame together as one > > entity to solve the issue. > > > > Bug: webrtc:8773 > > Change-Id: Ibe746a46cb41fd92b399a7069e1d89f02f292af7 > > Reviewed-on: https://webrtc-review.googlesource.com/38481 > > Commit-Queue: Qiang Chen <qiangchen@chromium.org> > > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21737} > > TBR=qiangchen@chromium.org,emircan@webrtc.org > > Change-Id: I11eff814ce093bf6db327ebcd21b1b71a1929849 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8773 > Reviewed-on: https://webrtc-review.googlesource.com/43260 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21739} TBR=deadbeef@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org Change-Id: I0d64b7e7a62e4f35aa012270d3826a23b3fb2337 Bug: webrtc:8773 Reviewed-on: https://webrtc-review.googlesource.com/43440 Commit-Queue: Qiang Chen <qiangchen@chromium.org> Reviewed-by: Qiang Chen <qiangchen@chromium.org> Cr-Commit-Position: refs/heads/master@{#21749}
2018-01-24 13:36:41 -08:00
"../rtp_rtcp:rtp_rtcp_format",
Add StereoCodecAdapter classes This CL is the step 1 for adding alpha channel support over the wire in webrtc. - Add the footprint for adapter classes that wraps actual codecs. - This CL does not add a webrtc::VideoFrame container that can carry alpha to make the CL shorter for an easier review. Therefore, it exercises a code path for when we receive no alpha input, just regular I420 frames. - Unittest sends a video frame for encode/decode through these adapters and checks the output PSNR. - See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental CL that gives an idea about how it will come together. Design Doc: https://goo.gl/sFeSUT Bug: webrtc:7671 Change-Id: I9d3be13647a0a958feceb8d7a9aa93852fc6a1fa Reviewed-on: https://webrtc-review.googlesource.com/11841 Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20490}
2017-10-30 23:10:12 -07:00
]
}
Reland "Reland "Reland "Put internal video codec factories into separate target""" This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff Original change's description: > Reland "Reland "Put internal video codec factories into separate target"" > > This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26 > Original change's description: > > Reland "Put internal video codec factories into separate target" > > > > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258 > > Original change's description: > > > Put internal video codec factories into separate target > > > > > > The purpose is to start splitting out the dependencies to the built-in > > > SW video codecs, so that clients can decide to not depend on them and > > > get a reduction in binary size. > > > > > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101 > > > > > > Bug: webrtc:7925 > > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c > > > Reviewed-on: https://webrtc-review.googlesource.com/33420 > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21381} > > > > Bug: webrtc:7925 > > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842 > > Reviewed-on: https://webrtc-review.googlesource.com/35261 > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21389} > > Bug: webrtc:7925 > Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754 > Reviewed-on: https://webrtc-review.googlesource.com/35501 > Commit-Queue: Anders Carlsson <andersc@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21464} Bug: webrtc:7925 Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454 Reviewed-on: https://webrtc-review.googlesource.com/37000 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-04 15:10:22 +01:00
# This target includes the internal SW codec.
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",
Hide libvpx vp8 encoder behind an interface and add mock for testing. Bug: webrtc:9809 Change-Id: I27baa0309511cbd849c09c5d063a64d1fb1fcebf Reviewed-on: https://webrtc-review.googlesource.com/c/103442 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25005}
2018-10-04 17:52:36 +02:00
"codecs/vp8/libvpx_interface.cc",
"codecs/vp8/libvpx_interface.h",
Split vp8_impl into webm_vp8_encoder and webm_vp8_decoder This work is in preparation for refactoring the TemporalLayers api. Bug: webrtc:9012 Change-Id: I01908ee034fb79996e687ff72d10178acf102321 Reviewed-on: https://webrtc-review.googlesource.com/61781 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22445}
2018-03-14 17:52:55 +01:00
"codecs/vp8/libvpx_vp8_decoder.cc",
"codecs/vp8/libvpx_vp8_decoder.h",
"codecs/vp8/libvpx_vp8_encoder.cc",
"codecs/vp8/libvpx_vp8_encoder.h",
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",
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",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Fix style violations in common_types.h and config.h Mostly, it's about moving constructors and descructors to the .cc files, so that they won't be inlined everywhere. The reason this CL is so big is that a lot of code was using common_types.h without declaring a dependency on webrtc_common, which broke the build once common_types.h started to depend on common_types.cc. BUG=163 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26089004 Cr-Commit-Position: refs/heads/master@{#8516} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-26 13:59:22 +00:00
"../..:webrtc_common",
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",
[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_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
libvpx-vp8: Add settings struct to constructor. Migrate the injectable Vp8FrameBufferControllerFactory into a settings struct, allowing for straight-forward future extensions. Bug: webrtc:11436 Change-Id: I53e555eb6ef88cf5b10ee8a43abd6ef9c930d100 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170635 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30839}
2020-03-20 09:34:32 +01:00
"../../rtc_base:deprecation",
Remove remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
vp8_impl.cc: Make it possible to base postproc deblocking level for arm on qp (e.g. turn off deblocking for low qp values). BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2970923002 Cr-Commit-Position: refs/heads/master@{#19300}
2017-08-09 23:37:05 -07:00
"../../rtc_base:rtc_numerics",
Add 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",
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",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Removing $rtc_libyuv_dir and removing useless dependencies on libyuv. This CL removes the following GN variables: rtc_build_libyuv, rtc_libyuv_dir (as requested in webrtc:7906). It also removes some unneeded dependencies on //third_party/libyuv. WebRTC targets were using public_deps to depend on //third_party/libyuv and this created a build graph where targets that were depending on //third_party/libyuv were not declaring the dependency to GN because they were somehow getting it from another target that was exposing //third_party/libyuv header files even if it wasn't directly depending on it. Bug: webrtc:8605, webrtc:7906 Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb Reviewed-on: https://webrtc-review.googlesource.com/32201 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:24:00 +01:00
"//third_party/libyuv",
Add initial gn build files for video_coding and video_processing. BUG= R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
GN: Enable libvpx, add link target and convert some test targets Libvpx now supports GN and this CL turns on compiling it. I also introduced an executable target 'webrtc_tests' that depends on all in WeBRTC + tests in order to get a full linking step executed (since we've seen link problems for GN when rolling WebRTC into Chromium). I also converted a few test targets and made a GN file for third_party/gflags. BUG=3441 TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc dir to a workspace wit this CL applied. R=brettw@chromium.org TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25569004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
}
GN: Add BUILD.gn files + kjellander to OWNERS This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
}
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
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",
"..:module_api",
"../..:webrtc_common",
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",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_numerics",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"codecs/vp9/svc_rate_allocator.cc",
"codecs/vp9/svc_rate_allocator.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",
"../..:webrtc_common",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api/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",
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",
"//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",
"codecs/vp9/vp9.cc",
"codecs/vp9/vp9_frame_buffer_pool.cc",
"codecs/vp9/vp9_frame_buffer_pool.h",
"codecs/vp9/vp9_impl.cc",
"codecs/vp9/vp9_impl.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",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ ) Reason for revert: Fixing the Gn error and try to reland. Original issue's description: > Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) > > Reason for revert: > Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio > > Original issue's description: > > Creating webrtc/modules:module_api > > > > This target keeps track of .h the files under webrtc/modules/include/ > > that are not part of any target. > > If a .h file is not part of a target the 'gn check' utility is not > > able to spot if a target is missing a dependency because even if > > it parses '#include' directives it is not able to find a target that > > contains these headers. > > > > BUG=webrtc:7513 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2838873002 > > Cr-Commit-Position: refs/heads/master@{#17880} > > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7513 > > Review-Url: https://codereview.webrtc.org/2839963005 > Cr-Commit-Position: refs/heads/master@{#17881} > Committed: https://chromium.googlesource.com/external/webrtc/+/bb08c3e29656fafe8a2d5d16ec4a62db49689f8a TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=kjellander@webrtc.org BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2843913002 Cr-Commit-Position: refs/heads/master@{#17884}
2017-04-26 03:38:35 -07:00
"..:module_api",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
"../..:webrtc_common",
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",
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",
"../../media:rtc_vp9_profile",
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",
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",
Configure frame references in VP9 encoder wrapper. Bug: webrtc:9585 Change-Id: I3f90d8f2b81556cfb5fa9123607ab0a9ade2bf3f Reviewed-on: https://webrtc-review.googlesource.com/93469 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24915}
2018-10-01 17:13:50 +02:00
"../../system_wrappers:field_trial",
Framerate controller for VP9 screen sharing. - Limit framerate by dropping frames before encoding. - The max framerate at screen sharing is set to 5fps. Bug: webrtc:9261 Change-Id: Icfbbecce33fdce2d746291708db0108e0ba10760 Reviewed-on: https://webrtc-review.googlesource.com/76921 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23316}
2018-05-18 17:31:19 +02:00
"../rtp_rtcp:rtp_rtcp_format",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
]
if (rtc_build_libvpx) {
PRESUBMIT: Improve PyLint check and add GN format check. Add pylintrc file based on https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc bit tightened up quite a bit (the one in depot_tools is far more relaxed). Remove a few excluded directories from pylint check and fixed/ suppressed all warnings generated. Add GN format check + formatted all GN files using 'gn format'. Cleanup redundant rules in tools/PRESUBMIT.py TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations. Ran it again with a modification in webrtc/build/webrtc.gni, formatted all the GN files and ran it again. R=henrika@webrtc.org, phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50069004 Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 12:55:39 +02:00
deps += [ rtc_libvpx_dir ]
Add VP9 codec to VCM and vie_auto_test. Include VP9 tests in videoprocessor_integrationtests. Include end-to-end send/receiveVP9 test. This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in: see https://code.google.com/p/webrtc/issues/detail?id=3932 R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
}
}
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
if (rtc_include_tests) {
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
if (is_android) {
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",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
"../../modules:module_api",
Update videoprocessor tool to use new video factory interface This will also cause us to use the new Android HardwareVideoEncoder, instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately, the new HW encoder does not seem to work as good as the old (or the new encoder is more strict with return values or something). I don't think it adds much value to continue testing the deprecated encoder, so I filed a bug for fixing the new encoder, and in this CL I disabled the tests on Android. I want to remove as many places as possible where we use the old WebRtcVideoEncoderFactory interface, because it makes it more difficult to migrate to the new interface. Bug: webrtc:7925 Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9 Reviewed-on: https://webrtc-review.googlesource.com/27540 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21037}
2017-12-04 14:09:40 +01:00
"../../rtc_base:rtc_base_approved",
Create experimental Obj-C++ API. This can be used to wrap Objective-C components in C++ classes, so users can use the WebRTC C++ API directly together with the iOS specific components provided by our SDK. Bug: webrtc:8832 Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8 Reviewed-on: https://webrtc-review.googlesource.com/46162 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21850}
2018-02-01 15:47:05 +01:00
"../../sdk:native_api",
Fix issues found by gn check. It turns out that some headers were not owned by any targets. These were: RTCVideoCodec.h RTCVideoCodecFactory.h RTCVideoCodecH264.h RTCVideoEncoderVP8.h RTCVideoDecoderVP8.h RTCVideoEncoderVP9.h RTCVideoDecoderVP9.h And some were owned by multiple targets, namely: RTCPeerConnectionFactory+Native.h RTCPeerConnectionFactory+Private.h RTCVideoFrameBuffer.h These have all been moved to their appropriate homes. This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files. Bug: webrtc:8855 Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7 Reviewed-on: https://webrtc-review.googlesource.com/49060 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22052}
2018-02-16 10:49:22 +01:00
"../../sdk:peerconnectionfactory_base_objc",
Re-enabling 'gn check': modules/video_coding:objc_codec_factory_helper. Bug: webrtc:8850 Change-Id: Ia00270b01fc143d470c5e814e4f31dfe2ce1fe78 Reviewed-on: https://webrtc-review.googlesource.com/54315 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22086}
2018-02-19 09:58:00 +01:00
"../../sdk:videocodec_objc",
Fix native api in preparation for native_api example. Add native api conversions for video frames and video renderer. This also requires some changes to sdk/BUILD to avoid cyclic dependencies. Bug: webrtc:8832 Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790 Reviewed-on: https://webrtc-review.googlesource.com/57142 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22340}
2018-03-07 10:32:03 +01:00
"../../sdk:videosource_objc",
Rename all objc targets to be suffixed for consistency BUG=webrtc:6412 Review-Url: https://codereview.webrtc.org/3001023003 Cr-Commit-Position: refs/heads/master@{#19489}
2017-08-24 05:43:45 -07:00
"../../sdk:videotoolbox_objc",
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
]
}
}
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",
Refactor SimulcastTestUtility into SimulcastTestFixture{,Impl} This will allow exposing the interface to downstream users that want to test VP8 simulcast. No functional changes to the tests themselves are expected. Bug: webrtc:9281 Change-Id: I4128b8f35a4412c5b330cf55c8dc0e173d4570da Reviewed-on: https://webrtc-review.googlesource.com/77361 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23469}
2018-05-31 12:53:00 +02:00
"../../:webrtc_common",
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",
"../../api/video:video_frame_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
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",
"../../api/video:video_frame_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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 remains of webrtc/base All downstream code have been updated to the new location. In PRESUBMIT.py: * Remove webrtc/rtc_base from CPP_BLACKLIST * Add webrtc/rtc_base to LEGACY_API_DIRS Fix some duplicated paths in webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn BUG=webrtc:7634 TBR=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2976293002 Cr-Commit-Position: refs/heads/master@{#19094}
2017-07-19 10:40:47 -07:00
"../../rtc_base:rtc_base_approved",
VideoProcessor: make it runnable on a task queue. * Guard members with a SequencedTaskChecker. * Intercept encoder/decoder callbacks, and post onto task queue if needed. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2996253002 Cr-Commit-Position: refs/heads/master@{#19428}
2017-08-21 06:44:16 -07:00
"../../rtc_base:rtc_task_queue",
Replacing SequencedTaskChecker with SequenceChecker. Bug: webrtc:9883 Change-Id: I5e3189da2a46e6f4ed1a3c5a5dfd2f7d75a16b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130961 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27518}
2019-04-09 13:44:04 +02:00
"../../rtc_base/synchronization:sequence_checker",
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",
[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",
Adding simulcast/spatial layering support to VideoProcessor. Encoded frames are preserved and decoded after all layers are encoded. Each spatial layer is decoded with separate decoder. For quality evaluation of lowres layers original input frame is downscaled with bilinear interpolation. Encoded and decoded frames are dumped into separate files. For async codecs encoded frames are passed to decoder in encode callback, as before. Bug: webrtc:8524 Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1 Reviewed-on: https://webrtc-review.googlesource.com/43381 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21844}
2018-02-01 13:25:17 +01:00
"//third_party/libyuv",
GN: Add modules_unittests Changes: * Enabled protobuf for iOS globally. * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope similar to GYP since tests depend on it. * Added missing rtc_libvpx_build_vp9 variable. * Moved out audio_coding defines into .gni file to avoid code duplication * Renamed files to avoid object naming conflicts that GN disallows: * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc} * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc} BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. Also ran: gn gen out/Default --args="rtc_enable_bwe_test_logging=true" and verified that more objects are being built (1885 vs 1883) when compiling modules_unittests. NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2041233006 Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 00:19:48 -07:00
]
}
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
Use frame generator in video codec unit tests. There is no need to use real video as input for encoder in unit tests. Using generator simplifies testing on mobile devices (no need to upload files to device). Bug: none Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398 Reviewed-on: https://webrtc-review.googlesource.com/64526 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:04:35 +02:00
video_coding_modules_tests_resources = []
Add MediaCodec VP tests for uncommon resolutions. Bug: None Change-Id: Ibfc35af3635c3b3a50027c4cd828f78e7a438dcd Reviewed-on: https://webrtc-review.googlesource.com/72342 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23020}
2018-04-25 12:45:16 +02:00
if (is_android) {
video_coding_modules_tests_resources += [
"../../resources/foreman_128x96.yuv",
"../../resources/foreman_160x120.yuv",
"../../resources/foreman_176x144.yuv",
"../../resources/foreman_240x136.yuv",
"../../resources/foreman_320x240.yuv",
"../../resources/foreman_480x272.yuv",
]
}
Enable more VideoCodecTests on iOS. Bug: webrtc:4755 Change-Id: I403834dbe04cc3899847eb10e5595a24e6001507 Reviewed-on: https://webrtc-review.googlesource.com/76602 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23273}
2018-05-16 16:46:58 +02:00
if (!is_android) {
Cleaning up modules_tests resources. * Removed video files which were not used by any tests. * Removed ConferenceMotion_1280_720_50.yuv for mobile builds. Bug: webrtc:8936 Change-Id: I0539e9fce20470fcc2f0af84bd297faffc4b587a Reviewed-on: https://webrtc-review.googlesource.com/60942 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22359}
2018-03-09 11:04:15 +01:00
video_coding_modules_tests_resources += [
"../../resources/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",
"../..:webrtc_common",
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2. This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/106280. This time the whole code base is covered. Some files may have not been fixed though, whenever the IWYU tool was breaking the build. Bug: webrtc:8311 Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef Reviewed-on: https://webrtc-review.googlesource.com/c/111965 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 16:47:49 +01:00
"../../api:array_view",
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: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",
"../../media:rtc_h264_profile_id",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../media:rtc_internal_video_codecs",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../media:rtc_media_base",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../rtc_base:checks",
Remove adapter bools from VideoCodecTestFixture::Config. It should be the responsibility of the fixture user to provide the exact codecs that should be tested instead. This reduces the coupling between the test fixture and the codec instantiation. Bug: webrtc:9317 Change-Id: I60d8f5c4b516ba33e2293d574ba17602c39f992b Reviewed-on: https://webrtc-review.googlesource.com/79147 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23425}
2018-05-29 09:31:15 +02:00
"../../rtc_base:rtc_base_approved",
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
"../../rtc_base:rtc_base_tests_utils",
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",
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",
[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",
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",
"../../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",
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",
Hide libvpx vp8 encoder behind an interface and add mock for testing. Bug: webrtc:9809 Change-Id: I27baa0309511cbd849c09c5d063a64d1fb1fcebf Reviewed-on: https://webrtc-review.googlesource.com/c/103442 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25005}
2018-10-04 17:52:36 +02:00
"codecs/vp8/test/mock_libvpx_interface.h",
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
"codecs/vp8/test/vp8_impl_unittest.cc",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
"codecs/vp9/test/vp9_impl_unittest.cc",
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
]
Move H.264 codec test activation directly behind build flag. Bug: None Change-Id: I5dd366aba16b440a8e99826e116db0d6b68bf844 Reviewed-on: https://webrtc-review.googlesource.com/76621 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23231}
2018-05-14 17:11:49 +02:00
if (rtc_use_h264) {
sources += [ "codecs/test/videocodec_test_openh264.cc" ]
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
deps = [
[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",
Reland "Rename stereo video codec to multiplex" This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174. Original change's description: > Rename stereo video codec to multiplex > > This CL only does the rename from"stereo" to multiplex". With this we have a > better name that doesn't clash with audio's usage of stereo. > > Bug: webrtc:7671 > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652 > Reviewed-on: https://webrtc-review.googlesource.com/43242 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21769} TBR=niklas.enbom@webrtc.org Bug: webrtc:7671 Change-Id: I6f38dc46126f279f334d52b56339b40acdc30511 Reviewed-on: https://webrtc-review.googlesource.com/45820 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21794}
2018-01-25 13:01:09 -08:00
":webrtc_multiplex",
Generalize SimulcastEncoderAdapter, use for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 16:16:38 +02:00
":webrtc_vp8",
Add QP for libvpx VP9 decoder. Also adds a basic unit test for VP9 implementation. BUG=webrtc:6541 Review-Url: https://codereview.webrtc.org/2654813002 Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 02:54:57 -08:00
":webrtc_vp9",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
":webrtc_vp9_helpers",
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
"../..:webrtc_common",
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",
New file api/video/BUILD.gn Build targets involving files under api/video/ are moved into this file, from api/BUILD.gn. In addition, drop "_api" part of target names, and move the header file api/videosinkinterface.h to api/video/video_sink_interface.h. Bug: webrtc:9253 Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155 Reviewed-on: https://webrtc-review.googlesource.com/75500 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-11 11:15:30 +02:00
"../../api/video:video_frame_i420",
Prepare for splitting the api/video:video_frames build rule. This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files. Bug: webrtc:10668 Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291 Commit-Queue: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 12:59:38 +02:00
"../../api/video:video_rtp_headers",
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",
Stop using public_deps in media/. Bug: webrtc:8603 Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/33180 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-11 17:07:30 +01:00
"../../media:rtc_h264_profile_id",
Fix build errors when rtc_use_builtin_sw_codecs is set to false. The previous effort of building WebRTC without SW codecs stopped when libjingle_peerconnection was possible to build. In order to make the group("default") target build, this basically updates a bunch of tests to explicitly depend on the built-in software video codecs. Bug: webrtc:7925 Change-Id: I2715414770c197fca01cb8dbde173a21f4434500 Reviewed-on: https://webrtc-review.googlesource.com/70503 Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23216}
2018-05-14 10:11:42 +02:00
"../../media:rtc_internal_video_codecs",
Add support for H.264 constrained high profile in VideoProcessor. BUG=webrtc:8448 Change-Id: I968d6cd78dd4f3c19a7944ae4cc73c5eddb9a949 Reviewed-on: https://webrtc-review.googlesource.com/16160 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20466}
2017-10-27 14:51:46 +02:00
"../../media:rtc_media_base",
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",
Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder"" This is a reland of cb853c8f90d3410a7f0ce07915aa20db0329259d Original change's description: > Reland "Add Profile 2 configuration to VP9 Encoder and Decoder" > > This is a reland of fc9c4e88b5569f0d2cd1c64cbc27fd969ce2db17 > > Original change's description: > > Add Profile 2 configuration to VP9 Encoder and Decoder > > > > Bug: webrtc:9376 > > Change-Id: I4f627fb2b6c146a90cfcaa815da459b09dc00003 > > Reviewed-on: https://webrtc-review.googlesource.com/81980 > > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Jerome Jiang <jianj@google.com> > > Cr-Commit-Position: refs/heads/master@{#23917} > > Bug: webrtc:9376 > Change-Id: I21fc44865af4e381f99dbc5ae2baf4a53ce834ca > TBR: niklas.enbom@webrtc.org > Reviewed-on: https://webrtc-review.googlesource.com/88341 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23974} TBR: niklas.enbom@webrtc.org Bug: webrtc:9376 Change-Id: I90d7ebc2110b82901656df7f9331ae82ee010baf Reviewed-on: https://webrtc-review.googlesource.com/88582 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23977}
2018-07-13 17:10:00 -07:00
"../../media:rtc_vp9_profile",
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",
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",
Replace rtc::Optional with absl::optional This is a no-op change because rtc::Optional is an alias to absl::optional This CL generated by running script from modules with parameters 'pacing video_coding congestion_controller remote_bitrate_estimator': find $@ -type f \( -name \*.h -o -name \*.cc \) \ -exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \ -exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \ -exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+ find $@ -type f -name BUILD.gn \ -exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+; git cl format Bug: webrtc:9078 Change-Id: I8ea501d7f1ee36e8d8cd3ed37e6b763c7fe29118 Reviewed-on: https://webrtc-review.googlesource.com/83900 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23640}
2018-06-18 10:48:16 +02:00
"//third_party/abseil-cpp/absl/types:optional",
Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder"" This is a reland of cb853c8f90d3410a7f0ce07915aa20db0329259d Original change's description: > Reland "Add Profile 2 configuration to VP9 Encoder and Decoder" > > This is a reland of fc9c4e88b5569f0d2cd1c64cbc27fd969ce2db17 > > Original change's description: > > Add Profile 2 configuration to VP9 Encoder and Decoder > > > > Bug: webrtc:9376 > > Change-Id: I4f627fb2b6c146a90cfcaa815da459b09dc00003 > > Reviewed-on: https://webrtc-review.googlesource.com/81980 > > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Jerome Jiang <jianj@google.com> > > Cr-Commit-Position: refs/heads/master@{#23917} > > Bug: webrtc:9376 > Change-Id: I21fc44865af4e381f99dbc5ae2baf4a53ce834ca > TBR: niklas.enbom@webrtc.org > Reviewed-on: https://webrtc-review.googlesource.com/88341 > Commit-Queue: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Emircan Uysaler <emircan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23974} TBR: niklas.enbom@webrtc.org Bug: webrtc:9376 Change-Id: I90d7ebc2110b82901656df7f9331ae82ee010baf Reviewed-on: https://webrtc-review.googlesource.com/88582 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23977}
2018-07-13 17:10:00 -07:00
"//third_party/libyuv",
Reland of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #1 id:1 of https://codereview.webrtc.org/2651023005/ ) Reason for revert: Will change the name from bwe_simulator to bwe_simulator_test. Original issue's description: > Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ ) > > Reason for revert: > Did break the bots. > https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807 > > Original issue's description: > > GN: Refactor modules_unittests to eliminate package boundary violations. > > > > Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator > > > > BUG=webrtc:6954 > > NOTRY=True > > > > Review-Url: https://codereview.webrtc.org/2649563002 > > Cr-Commit-Position: refs/heads/master@{#16270} > > Committed: https://chromium.googlesource.com/external/webrtc/+/36cb55d7157dcba872f666e0e1fb75cae1a940a6 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6954 > > Review-Url: https://codereview.webrtc.org/2651023005 > Cr-Commit-Position: refs/heads/master@{#16271} > Committed: https://chromium.googlesource.com/external/webrtc/+/3373eaa57757386823730e9998cd225dd6be477c R=kjellander@webrtc.org NOTRY=true BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2653173003 Cr-Commit-Position: refs/heads/master@{#16356}
2017-01-30 03:07:03 -08:00
]
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
data = video_coding_modules_tests_resources
ObjC: Implement HW codecs in ObjC instead of C++ The current ObjC HW encoder is implemented as a C++ webrtc::VideoEncoder. We then wrap it two times in the following way: webrtc::VideoEncoder -> RTCVideoEncoder -> webrtc::VideoEncoder. This was originally done to minimize the code diff when landing the injectable encoder. This CL removes the first wrapping and implements the ObjC HW encoder as a RTCVideoEncoder directly. Similarly, the decoder is implemented as a RTCVideoDecoder directly. Based on andersc@ CL: https://codereview.webrtc.org/2978623002/. BUG=webrtc:7924 Review-Url: https://codereview.webrtc.org/2987413002 Cr-Commit-Position: refs/heads/master@{#19255}
2017-08-07 06:55:28 -07:00
Add VideoProcessorIntegrationTest for MediaCodec implementations. * Add VideoProcessorIntegrationTestMediaCodec tests, including a standard foreman test and a forced SW fallback test. * Morph PlotVideoProcessorIntegrationTest into VideoProcessorIntegrationTestParameterized. This test is intended to be patched locally, depending on what metric is of interest. It is run on the bots to ensure that it doesn't break. * Remove the plot_videoprocessor_integrationtest binary. The test above is instead moved to the modules_tests binary. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/3011043002 Cr-Commit-Position: refs/heads/master@{#19727}
2017-09-07 07:50:06 -07:00
if (is_android) {
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
sources += [ "codecs/test/videocodec_test_mediacodec.cc" ]
Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)." This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e. Reason for revert: On Lollipop Nexus 4, the 240p tests fail too. Original change's description: > Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld). > > * Add audio_ prefix to CallTest::{en,de}coder_factory_. > * Let VideoQualityTest only instantiate encoders using encoder factories. > * Add HW encoder factories to VideoQualityTest. > * Add full stack tests: > - sqcif7 at 30 kbps: libvpx. > - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx. > > BUG=webrtc:8219 > TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org > > Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74 > Reviewed-on: https://webrtc-review.googlesource.com/4740 > Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20041} TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8219 Reviewed-on: https://webrtc-review.googlesource.com/4900 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:12 +00:00
Move all Android specific stuff to android_codec_factory_helper. This helps separate concerns, so that the VideoProcessorIntegrationTest is almost oblivious to the fact that it needs to connect to the JVM to get the Android HW codecs. Bug: webrtc:8448 Change-Id: I4359b31f84be48eaf99d83525bcce6e593e874f8 Reviewed-on: https://webrtc-review.googlesource.com/47384 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21890}
2018-02-05 09:52:31 +01:00
deps += [ ":android_codec_factory_helper" ]
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
}
if (is_ios || is_mac) {
Rewrite videoprocessor integrationtest to use public fixture. This CL creates a test fixture for the videoprocessor integration tests and exposes it as part of the public API. It also rewrites the current versions of the tests to build on this new paradigm. The motivation for this is to easily allow projects that build on top of webrtc to add integration-level tests for their own custom codec implementations in a way that does not link them too tightly to the internal implementations of said tests. Bug: None Change-Id: I7cf9f29322a6934b3cfc32da02ea7dfa5858c2b2 Reviewed-on: https://webrtc-review.googlesource.com/72481 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23118}
2018-05-04 11:56:55 +02:00
sources += [ "codecs/test/videocodec_test_videotoolbox.cc" ]
Add VideoProcessor tests verifying that H.264 keyframes contain SPS/PPS/IDR. This CL adds an EncodedFrameChecker interface which can be used by users of the VideoProcessor to inject customized per-frame checks to the encoding/decoding pipeline. This currently has two uses: - Verifying that the QP parser works correctly for VP8 and VP9, by comparing the parsed QP to that produced by libvpx. - Verifying that our H.264 encoders always produce SPS/PPS/IDR in tandem. TESTED=Galaxy S8, Pixel 2 XL, iPhone 7. BUG=webrtc:8423 Change-Id: Ic3e401546e239a9ffaf2ed2907689cebb1127805 Reviewed-on: https://webrtc-review.googlesource.com/14559 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20409}
2017-10-24 10:16:33 +02:00
Use frame generator in video codec unit tests. There is no need to use real video as input for encoder in unit tests. Using generator simplifies testing on mobile devices (no need to upload files to device). Bug: none Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398 Reviewed-on: https://webrtc-review.googlesource.com/64526 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22648}
2018-03-28 13:04:35 +02:00
deps += [ ":objc_codec_factory_helper" ]
if (num_video_coding_modules_tests_resources > 0) {
deps += [ ":video_coding_modules_tests_resources_bundle_data" ]
}
Add support for creating HW codecs in the VideoProcessor tests. This CL adds the ability to _create_ HW codecs (Android and iOS) in the VideoProcessor integration tests. Since the VideoProcessor class is not thread safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A follow-up CL is planned that will add this ability. This CL further adds a separate build target which is used to separate the "plot" versions of the integration tests from the "correctness" versions. The former will be run manually on devices, whereas the latter are used on the trybots/buildbots to find regressions in the SW codecs. The underlying test is the same, however. BUG=webrtc:6634 Review-Url: https://codereview.webrtc.org/2695653002 Cr-Commit-Position: refs/heads/master@{#16716}
2017-02-20 04:35:52 -08:00
}
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",
Adding layering configurator and rate allocator for VP9 SVC. The configurator decides number of spatial layers, their resolution and bitrate thresholds based on given input resolution and maximum number of spatial layers. The allocator distributes available bitrate across spatial and temporal layers. If there is not enough bitrate to provide acceptable quality for all spatial layers allocator disables enhancement layers one by one until the condition is met or number of layers is reduced to one. VP9 SVC related unit tests have been updated. Input resolution and bitrate in these tests have been increased to the level enough to provide desirable number of spatial layers. Bug: webrtc:8518 Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b Reviewed-on: https://webrtc-review.googlesource.com/60340 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22672}
2018-03-28 19:32:37 +02:00
"codecs/vp9/svc_rate_allocator_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"decoding_state_unittest.cc",
Make fec controller plug-able. Bug: webrtc:8656 Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592 Reviewed-on: https://webrtc-review.googlesource.com/39760 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-19 17:58:57 +01:00
"fec_controller_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"frame_buffer2_unittest.cc",
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",
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",
"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",
Fork NackModule and RtpVideoStreamReceiver Bug: webrtc:11595 Change-Id: I4d14c0bf9c32e09d1624099a256f2778afebd4df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175901 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31337}
2020-05-22 17:36:36 +02:00
"nack_module2_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",
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",
"session_info_unittest.cc",
"test/stream_generator.cc",
"test/stream_generator.h",
"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",
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 VP9 frame rate controller to separate class. Bug: webrtc:9669 Change-Id: I6f30587778e9783182af11d2410464024918e171 Reviewed-on: https://webrtc-review.googlesource.com/96201 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24487}
2018-08-29 01:05:26 +02:00
"utility/framerate_controller_unittest.cc",
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",
"utility/quality_scaler_unittest.cc",
"utility/simulcast_rate_allocator_unittest.cc",
Reland of Add experimental simulcast screen content mode The original CL was reverted because of a bug discovered by the chromium bots. Description of that CL: > Review-Url: https://codereview.webrtc.org/2636443002 > Cr-Commit-Position: refs/heads/master@{#16135} > Committed: https://chromium.googlesource.com/external/webrtc/+/a28e971e3bc8a06f366f0ef82cc7168b05a83b59 The first patch set of this CL is the same as r16135. Subsequence patch sets are the fixes applied. Some new test cases have been added, which reveal a few more bugs that have also been fixed. BUG=webrtc:4172 Review-Url: https://codereview.webrtc.org/2641133002 Cr-Commit-Position: refs/heads/master@{#16299}
2017-01-26 06:12:26 -08:00
"video_codec_initializer_unittest.cc",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
"video_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",
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",
Moved NackModule and VCMPacket to their own targets Bug: webrtc:9373 Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251 Reviewed-on: https://webrtc-review.googlesource.com/81744 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-07 16:15:40 +02:00
":nack_module",
Generalize SimulcastEncoderAdapter, use for H264 & VP8. * Move SimulcastEncoderAdapter out under modules/video_coding * Move SimulcastRateAllocator back out to modules/video_coding/utility * Move TemporalLayers and ScreenshareLayers to modules/video_coding/utility * Move any VP8 specific code - such as temporal layer bitrate budgeting - under codec type dependent conditionals. * Plumb the simulcast index for H264 in the codec specific and RTP format data structures. TBR=sprang@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org Bug: webrtc:5840 Change-Id: I2d3b130622dd7ceec5528f3ab6c46f109e6bafb8 Reviewed-on: https://webrtc-review.googlesource.com/84743 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23715}
2018-06-21 16:16:38 +02:00
":simulcast_test_fixture_impl",
Add untracked headers to video_coding. This creates a new target for pure defines and interfaces. I think that makes sense (though include/ makes it harder to see when .cc and .h files should live together). Bug: webrtc:7620 Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e Reviewed-on: https://webrtc-review.googlesource.com/34657 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21516}
2018-01-08 11:05:10 +01:00
":video_codec_interface",
GN: Refactor modules_unittests to eliminate package boundary violations. BUG=webrtc:6954 Review-Url: https://codereview.webrtc.org/2629923002 Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 08:27:11 -08:00
":video_codecs_test_framework",
":video_coding",
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",
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",
[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",
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",
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",
"../../api/video:video_frame_i420",
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 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",
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",
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",
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",
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",
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",
Add unittests to validate scalability structures without encoder Bug: webrtc:10342 Change-Id: I66407e635502b7c87f8d4ab49c95f5c1326da4a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176412 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31423}
2020-06-03 12:01:41 +02:00
"codecs/av1:scalability_structure_tests",
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",
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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