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WWebRTC LUCI CQFix AudioMixer histogram test
5574afc0创建于 2023年10月17日历史提交
文件最后提交记录最后更新时间
g3doc
Reland "Migrate WebRTC documentation to new renderer" This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > Reason for revert: > > Original change's description: > > Migrate WebRTC documentation to new renderer > > > > Bug: b/258408932 > > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39205} > > Bug: b/258408932 > Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39209} Bug: b/258408932 Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660 Owners-Override: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39231} 3 年前
BUILD.gn
Add histogram for audio mixer maximum source count. This adds the histogram WebRTC.Audio.AudioMixer.NewHighestSourceCount which logs the highest number of sources an AudioMixer has had. The statistic is logged whenever the highest number of sources increases. This allows us to differentiate the statistic to see how many times the mixer has had a certain maximum number of sources. Chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4414896 Bug: chromium:1430806 Change-Id: Iab92e201a0b667741cc8f3bbbed92fa989d7fcda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300860 Reviewed-by: Olga Sharonova <olka@webrtc.org> Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39833} 3 年前
DEPS
Enables usage of ChannelMixer in WebRTC's output mixer. Ensures that newly added ChannelMixer is utilized when number of channels is larger than two in the output mixer. Decided to land with henrik.lundin as TBR since he has reviewed all other changes in WebRTC related to channel mixing for multi-channel cases. All this CL does is to ensure that the new channel mixing scheme can be used in Chrome. The old scheme is still used for mono and stereo combinations. TBR: henrik.lundin Bug: webrtc:10783 Change-Id: I11c02f1b4ef60e847095efbcd5e5f5faf27a5cdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140290 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28517} 6 年前
OWNERS
Add conceptual documentation for Audio - Mixer NOTRY=true Bug: webrtc:12570 Change-Id: Iece5588c5a45a8619afb32c812ff671a161e48f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215929 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33806} 5 年前
audio_frame_manipulator.cc
Enables usage of ChannelMixer in WebRTC's output mixer. Ensures that newly added ChannelMixer is utilized when number of channels is larger than two in the output mixer. Decided to land with henrik.lundin as TBR since he has reviewed all other changes in WebRTC related to channel mixing for multi-channel cases. All this CL does is to ensure that the new channel mixing scheme can be used in Chrome. The old scheme is still used for mono and stereo combinations. TBR: henrik.lundin Bug: webrtc:10783 Change-Id: I11c02f1b4ef60e847095efbcd5e5f5faf27a5cdd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140290 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28517} 6 年前
audio_frame_manipulator.h
[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
audio_frame_manipulator_unittest.cc
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
audio_mixer_impl.cc
Mix audio from all sources. Removes the top 3 filtering based on frame energy. This behaviour is unexpected for many application developers and the platform should not have such arbitrary limitations. Developers can still implement top-N filtering using WebAudio or an SFU (recommended to increase scalability). Performance is not really a concern in this case since decoders on all receive streams are called regardless if they are mixed or not (assuming packets are received). This also fixes glitches caused by the current implementation since sources are not ramped out. Bug: chromium:1446655,webrtc:13818 Change-Id: I179a6d68d2517b94ff2d99ec269031a54e5099e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310180 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40349} 2 年前
audio_mixer_impl.h
Mix audio from all sources. Removes the top 3 filtering based on frame energy. This behaviour is unexpected for many application developers and the platform should not have such arbitrary limitations. Developers can still implement top-N filtering using WebAudio or an SFU (recommended to increase scalability). Performance is not really a concern in this case since decoders on all receive streams are called regardless if they are mixed or not (assuming packets are received). This also fixes glitches caused by the current implementation since sources are not ramped out. Bug: chromium:1446655,webrtc:13818 Change-Id: I179a6d68d2517b94ff2d99ec269031a54e5099e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310180 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40349} 2 年前
audio_mixer_impl_unittest.cc
Fix AudioMixer histogram test If the tests are run in a different order, the test might fail. We fix this by resetting the histogram data at the start of the test. Change-Id: I6fb349609842b55f416cf2ec8cd93d0b4328960e Bug: chromium:1430806 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323801 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Auto-Submit: Fredrik Hernqvist <fhernqvist@google.com> Cr-Commit-Position: refs/heads/main@{#40946} 2 年前
audio_mixer_test.cc
Adopt absl::string_view in modules/audio_mixer/ Bug: webrtc:13579 Change-Id: I40db95dcbd8e7bc7423144cf9066414d9f62fd2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271283 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37792} 3 年前
default_output_rate_calculator.cc
Move dynamic memory allocations of webrtc::AudioMixerImpl from RT thead (4 vector allocations removed) Bug: webrtc:12035,webrtc:12036 Change-Id: Ie0d734cd0016a27c57809af67187ceb97f92f233 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188621 Commit-Queue: Olga Sharonova <olka@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32441} 5 年前
default_output_rate_calculator.h
Use backticks not vertical bars to denote variables in comments for /modules/audio_mixer Bug: webrtc:12338 Change-Id: I88c0824451f1448590df0f57bb094d39dffece66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227093 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34623} 4 年前
frame_combiner.cc
Remove FrameCombiner stats Stop logging WebRTC.Audio.AudioMixer.* histograms. Bug: chromium:1308711, chromium:1328289 Change-Id: Iba1c89a112842c532d99900cd54aee7f38f759fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38651} 3 年前
frame_combiner.h
Remove unused enum FrameCombiner::LimiterType Bug: webrtc:7494 Change-Id: Ied1c9c37ccf1c57802df9d1d62f8de7790d2ee94 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291326 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39190} 3 年前
frame_combiner_unittest.cc
Remove unused enum FrameCombiner::LimiterType Bug: webrtc:7494 Change-Id: Ied1c9c37ccf1c57802df9d1d62f8de7790d2ee94 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291326 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39190} 3 年前
gain_change_calculator.cc
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
gain_change_calculator.h
[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
output_rate_calculator.h
Cleanup after sorting out dependencies of OutputRateCalculator Bug: webrtc:12035,webrtc:12036 Change-Id: I774f640a96b80e4942e4166f69475fe47f1bd0ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189801 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Olga Sharonova <olka@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32451} 5 年前
sine_wave_generator.cc
[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
sine_wave_generator.h
[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