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Add possibility to set MetricsSet metadata. Bug: b/266997275 Change-Id: I2c4fadcff7044a8c72ef7e46caf4eff398e29f91 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291700 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39237} 3 年前
Remove all usage of //rtc_base target Bug: webrtc:9838 Change-Id: If813dbb426b4dc848185b64c0349d03fa9c059f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290986 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39116} 3 年前
In PCLF remove ability to inject TaskQueueFactory and CallFactory Instead rely on TaskQueueFactory and Clock provided by the internal TimeController of the PCLF framework. Bug: webrtc:15574 Change-Id: I473e1f12ead97f866dbd45771ed5a59541c0c47c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325182 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41026} 2 年前
Add stream label to test video source for better debugablity and testability Bug: b/294812400 Change-Id: I830515b797100ca2dc0e68dd3b79d5a1bb4068da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316221 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40581} 2 年前
Delete ProcessThread creation from test TimeController as unused Bug: webrtc:7219 Change-Id: Ia34f24a804b8a1e06b089774e37cac6e6d749e82 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266366 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37311} 3 年前
Adds more owners to api/test Bug: None Change-Id: Ica95e15f8521274c41b475d8c39a0b27a50c7724 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196090 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32740} 5 年前
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
Add ability for audioproc_f to operate on any AudioProcessing object. This CL extends the WebRTC testing API to allow audioproc_f -based testing using a pre-created AudioProcessing object. This is an important feature to allow testing any AudioProcessing objects that are injected into WebRTC. Beyond adding this, the CL also changes the simulation code to operate on a scoped_refptr<AudioProcessing> object instead of a std::unique<AudioProcessing> object Bug: webrtc:5298 Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31319} 5 年前
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
Fix errors in new SessionDescriptionInterface mock and really compile it with CompileAllHeaders. Bug: webrtc:14594 Change-Id: I51b0364cbede0e1d614ee708fbc01580bda68d3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280223 Commit-Queue: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38465} 3 年前
Add support for NV12 frame generator Bug: b/240540204 Change-Id: Id2205e8bd0dfd59476dcd68c32c4981f98b51422 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278402 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38322} 3 年前
Add support for NV12 frame generator Bug: b/240540204 Change-Id: Id2205e8bd0dfd59476dcd68c32c4981f98b51422 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278402 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38322} 3 年前
Add infrastructure stats for network emulation layer Bug: b/240540204 Change-Id: I66dfd25775faa9d1bc7e75a932a36e8aa97c0f57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38613} 3 年前
Add infrastructure stats for network emulation layer Bug: b/240540204 Change-Id: I66dfd25775faa9d1bc7e75a932a36e8aa97c0f57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282320 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38613} 3 年前
Use classes from media_configuration.h instead of the ones in PeerConnectionE2EQualityTestFixture. Classes defined inside the class PeerConnectionE2EQualityTestFixture are replaced by the ones define in media_configuration.h. Change-Id: I1c025ff10aacf8cbc3df9bfa742a40622fe0807a Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281860 Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38568} 3 年前
Use classes from media_configuration.h instead of the ones in PeerConnectionE2EQualityTestFixture. Classes defined inside the class PeerConnectionE2EQualityTestFixture are replaced by the ones define in media_configuration.h. Change-Id: I1c025ff10aacf8cbc3df9bfa742a40622fe0807a Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281860 Commit-Queue: Jeremy Leconte <jleconte@google.com> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38568} 3 年前
Migrate test_main_lib on new global metrics API Bug: b/246095034 Change-Id: I99cd631cdae49ad1e0812f1204a6be4d6f43bc34 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276604 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38184} 3 年前
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209} 6 年前
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}7 年前
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491} 6 年前
Add video codec tester. This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them. The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities. Other differences: - Run encoding and decoding in separate threads - Run quality analysis in a separate thread - Reference frame buffering is moved into video source (which re-read frames from the file). - Make it possible to run decode-only tests This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now. Bug: b/261160916 Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38901} 3 年前
Add video codec tester. This tester is an improved version of VideoProcessor and VideoCodecTestFixture and will eventually replace them. The tester provides better separation between codecs and testing logic. Its knowledge about codecs is limited to frame encode/decode calls and frame ready callbacks. Instantiation and configuration of codecs are the test responsibilities. Other differences: - Run encoding and decoding in separate threads - Run quality analysis in a separate thread - Reference frame buffering is moved into video source (which re-read frames from the file). - Make it possible to run decode-only tests This CL is MVP implementation: it adds only 1 test (video_codec_test.cc, ConstantRate/EncodeDecodeTest) and the test is disabled for now. Bug: b/261160916 Change-Id: Ida24a2fca1b1496237fa695c812084877c76379f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283525 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38901} 3 年前
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209} 6 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209} 6 年前
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}7 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
Add rtc::make_ref_counted to api/ This cl adds a forwarding header, a build target, and migrates headers in api/ to use it. Moving actual implementation, will follow, in https://webrtc-review.googlesource.com/c/src/+/265390. Bug: webrtc:12701 Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37200} 3 年前
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}7 年前
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED. The new macros will replace the old one when old one's usage will be removed. The idea of the renaming to provide a clear signal that this is debug build only macros and will be stripped in the production build. Bug: webrtc:9065 Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35348} 4 年前
Make FrameGeneratorInterface::fps() pure virtual. Bug: b/269577953 Change-Id: I418d241fe966fa3a38b851aaa70aaf59ee03ca57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295261 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39407} 3 年前
Convert AsyncDnsResolver to use absl::AnyInvocable Bug: webrtc:12598 Change-Id: I0950231d6de7cf53116a573dcd97a3cf5514946c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318400 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40670} 2 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Extend mocks for public types Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782} 3 年前
[DataChannelInterface] Introduce DataChannelInterface::SendAsync() One problem with the existing Send() method is that it has a return value that is problematic for a fully async implementation. A second problem with Send() is that the return value is bool and not RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError. Also, start deprecating bool Send() in favor of void SendAsync() and adding network_safety_ flag for posting async operations to the network thread. This flag also takes over from the connected_to_transport_ which can now be removed. Bug: webrtc:11547, webrtc:13289 Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39817} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
RtpSenderInterface::SetEncoderSelector This cl/ adds a way of setting an EncoderSelector on a specific RtpSenderInterface. This makes it possible to easily use different EncoderSelector on different streams within the same or different PeerConnections. The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor. Iff a EncoderSelector is set on the RtpSender, it will take precedence over the VideoEncoderFactory::GetEncoderSelector. Bug: webrtc:14122 Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37150} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Extend mocks for public types Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782} 3 年前
Reland "Move injection of PacketSocketFactory from PC to PCF" This is a reland of commit 905c3a6c73d293882ef11942066ccda52a9e14d1 Change from previous attempt is between ps#1 and ps#2: Use PeerConnectionFactoryInterface::Options to clear the network_ignore_mask. Original change's description: > Move injection of PacketSocketFactory from PC to PCF > > Injection via PeerConnectionDependecies was broken, in not accepting > ownership of the injected object. > > Bug: webrtc:7447, webrtc:14204 > Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#37270} Bug: webrtc:7447, webrtc:14204 Change-Id: Ic78ebec2e88a8c44699015c8c7a44e137f44253a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265982 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37290} 3 年前
New PeerConnectionFactory::CreateVideoTrack with refcounted source Bug: webrtc:15017 Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39635} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Rename header extension API methods following spec updates from https://github.com/w3c/webrtc-extensions/pull/142 BUG=chromium:1051821 Change-Id: I1fd991a5024d38ac59ebe510ea1a48fd6f42d23b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296321 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39491} 3 年前
Extend mocks for public types Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782} 3 年前
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627} 3 年前
Fix errors in new SessionDescriptionInterface mock and really compile it with CompileAllHeaders. Bug: webrtc:14594 Change-Id: I51b0364cbede0e1d614ee708fbc01580bda68d3d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280223 Commit-Queue: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38465} 3 年前
Remove deprecated TransformableAudioFrameInterface::getHeader() method Fixed: chromium:1456628 Change-Id: I12ea08070578de846f042c64f2808b55de1603a8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315960 Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40555} 2 年前
Expose video mimeType for insertable streams which allows determining what codec (data format) is used. Chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4941907 Split from https://webrtc-review.googlesource.com/c/src/+/318283 to reduce CL size and avoid audio woes. BUG=webrtc:15579 Change-Id: I404107af526df3009c16d2a6148784fe87dfa807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323721 Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#41007} 2 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Remove EncodedFrame::MissingFrame and start removing Decode() param Remove EncodedFrame::MissingFrame, as it was always false in actual in-use code anyway, and remove usages of the Decode missing_frames param within WebRTC. Uses/overrides in other projects will be cleaned up shortly, allowing that variant to be removed from the interface. Bug: webrtc:15444 Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#40662} 2 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Revert "Add checks for api/test mocks to make sure they're complete" This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb. Reason for revert: Breaks upstream. Original change's description: > Add checks for api/test mocks to make sure they're complete > > Also unifies the mock inheritance if they inherited from a ref counted > interface: > - it should only inherit from the interface > - it should use make_ref_counted > > Bug: webrtc:14594 > Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423 > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38602} Bug: webrtc:14594 Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940 Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38604} 3 年前
Extend mocks for public types Extends the mocks for rtpreceiver rtpsender and videotrack. This change allows the external HangoutsKit client to remove its own mocks of rtc types. Bug: none Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jack Smith <jackdsmith@google.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38782} 3 年前
Remove the move constructor from NetEqState. This move constructor causes downstream issues, so it needs to be removed for now. Bug: webrtc:9667 Change-Id: Ic15bfdf6b392a95e05bf75bc2c1dd32ce132d32b Reviewed-on: https://webrtc-review.googlesource.com/99121 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24672}7 年前
Expose run function to NetEqSimulator Bug: webrtc:11005 Change-Id: I84f01536b40ba17e66877cdced194e05b882b5c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167537 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30405} 6 年前
Add config options for python plots and text log to the NetEq simulator API. This CL also introduces a helper function to perform the config conversion, which eliminates duplicate code. Bug: webrtc:10337 Change-Id: I162288f90ebac8f2f345356ec25f0805df698c67 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188121 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32399} 5 年前
Add config options for python plots and text log to the NetEq simulator API. This CL also introduces a helper function to perform the config conversion, which eliminates duplicate code. Bug: webrtc:10337 Change-Id: I162288f90ebac8f2f345356ec25f0805df698c67 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188121 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32399} 5 年前
Add setters to NetworkEmulationManager::SimulatedNetworkNode::Builder. Bug: b/294494713 Change-Id: I89130a4742da5f04680aa38721afcd7f91fb30ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314980 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40511} 2 年前
Add setters to NetworkEmulationManager::SimulatedNetworkNode::Builder. Bug: b/294494713 Change-Id: I89130a4742da5f04680aa38721afcd7f91fb30ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314980 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40511} 2 年前
New struct PeerNetworkDependencies Preparation to make landing of https://webrtc-review.googlesource.com/c/src/+/238660 easier. Bug: webrtc:13145 Change-Id: I314a53cc634f842e5df009d0802b214aa6f8728b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238663 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35403} 4 年前
Remove deprecated AddPeer method. Change-Id: Icd15dc4d7d79276734260fb11932d9ede8dbbf23 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283661 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38659} 3 年前
Move media configuration classes out of PeerConnectionE2EQualityTestFixture. The goal is to remove the dependency between PeerConfigurerImpl and PeerConnectionE2EQualityTestFixture so that PeerConfigurerImpl can be used in PeerConnectionE2EQualityTestFixture API. Change-Id: I29ae44b9d0e39075d0c395ff9d9f8d313be12176 Bug: webrtc:14627 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281740 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#38560} 3 年前
Reland "Add documentation, tests and simplify webrtc::SimulatedNetwork." This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2 Original change's description: > Add documentation, tests and simplify webrtc::SimulatedNetwork. > > This CL increases the test coverage for webrtc::SimualtedNetwork, adds > some more comments to the class and the interface it implements and > simplify the logic around capacity and delay management in the > simulated network. > > More CLs will follow to continue the refactoring but this is the > ground work to make this more modular in the future. > > Bug: webrtc:14525, b/243202138 > Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38388} Bug: webrtc:14525, b/243202138, b/256595485 Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38557} 3 年前
generateKeyframe: pass frame_types in bypass mode Passes frame_types to the underlying encoder in bypass mode. For libvpx this has no effect, for H264 this changes the behavior to allow generating keyframes on a per-layer basis. BUG=chromium:1354101 Change-Id: I26fc22d9e2ec4681a57ce591e9eafd0b1ec962b0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285083 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#38821} 3 年前
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
Set up a new rtc::Thread instance per test. Several tests leave pending tasks behind after executing, which may affect the state of subsequent tests. This CL isolates each test in the sense that a dedicated Thread instance is created per test and then pending tasks are flushed and the Thread instance deleted. Down the line we may want to improve on this and flag those tests that leave pending tasks/timers etc. Change-Id: Ibaf3719a9974c57ac2169edca0e2a06a9ea6c78f Bug: webrtc:11574 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175132 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31268} 5 年前
[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}7 年前
Rename done() into condition(), because it is actually condition in TimeController API Bug: None Change-Id: Ia3a742d1d2ad1238223f4da7ae843a8d22108ec5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174060 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31144} 5 年前
Delete ProcessThread creation from test TimeController as unused Bug: webrtc:7219 Change-Id: Ia34f24a804b8a1e06b089774e37cac6e6d749e82 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266366 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37311} 3 年前
[PCLF] Propagate relevant metadata to all metrics Bug: None Change-Id: Ifcb67a59b68cc3468dd06e932a2a3da7b40d9845 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281680 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38545} 3 年前
Log metrics even if test failed Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression. This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet. Bug: webrtc:14852 Change-Id: I967160069055036f93e595d328c4d5f1ca483be9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39840} 3 年前
Log metrics even if test failed Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression. This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet. Bug: webrtc:14852 Change-Id: I967160069055036f93e595d328c4d5f1ca483be9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39840} 3 年前
Dump codec input Add functionality for dumping encoder and decoder input to file in video codec test. Bug: b/261160916, webrtc:14852 Change-Id: I49a84a886d87903c601cf5c35bd723b6393c2a75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298051 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39626} 3 年前
[DVQA] Add a GetSenderPeerName method. Change-Id: I2b30510911865150881c116abc2f86be7821f34a Bug: b/277851637 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301280 Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39875} 3 年前
Add --dependency_descriptor flag to video_loopback. Bug: webrtc:14801 Change-Id: I8151f66ceb118a7abd40bbdc5bff71b5fdf66cb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289961 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38977} 3 年前
Refactor video codec testing stats This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters. VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl. Bug: b/261160916, webrtc:14852 Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39248} 3 年前
Report encode/decode latency Bug: none Change-Id: If36ee02ee762718b1c1b6f84cd22cb866ba0d51b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251863 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36024} 4 年前
Refactor video codec testing stats This CL introduces VideoCodecStats and VideoCodecStatsImpl which provide baseline functionalities for storing, slicing and aggregation of encoded and/or decoded video frame statistics. To facilitate metrics logging (not implemented yet), SamplesStatsCounter is used for stream parameters. VideoCodecStats/VideoCodecStatsImpl will replace existing VideoCodecTestStats/VideoCodecTestStatsImpl. Bug: b/261160916, webrtc:14852 Change-Id: I0f96ce1ed9be3aee2a702804612524676c9882fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291323 Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39248} 3 年前