Commit Graph

694 Commits

Author SHA1 Message Date
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
bddfa1d9de Prepare the code to inherit from chromium's mb.py (3rd attempt).
Basically try to make low_bandwidth_audio_perf_test target work with chromium's mb.

Bug: webrtc:13867
Change-Id: If111e2b0f4f82935062d6321f4d16cc72c77d572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256975
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36364}
2022-03-29 10:11:51 +00:00
1b51b11f2a Fix low_bandwidth_audio_perf_test gn group definition on windows.
This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/256862.
Current run have the below error:
https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket/8818456476112979761/+/u/generate_build_files/stdout

The code is copied from Chromium:
https://crsrc.org/c/chrome/test/BUILD.gn;l=4540;bpv=1;bpt=0;drc=f63ec863909167deb96888769776621767a99fa2

Bug: webrtc:13867
No-Try: True
Change-Id: Idf4d3bf4acde66d4c48ecc9b5aa2053c7a82f033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36358}
2022-03-28 16:02:20 +00:00
d7fdb95346 Remove typing detection
Motivation: never used.
Follow-up CL of https://webrtc-review.googlesource.com/c/src/+/250680.

Tested on a custom chromium build by making a video call and with
keyboard activity. The expected logs from the APM transient suppressor
sub-module were shown.

Bug: webrtc:11226
Change-Id: I4186994412dd8ba2e71ed8f9dcc9cf8f8e40fbd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250667
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36300}
2022-03-23 10:23:54 +00:00
a943e730b2 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 7/inf
Convert audio/ and collateral (audio encoder copy red).

Bug: webrtc:10335
Change-Id: Iac54c0cfd2f62f4402f3deec35ae2725ec35b81a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36229}
2022-03-17 07:11:44 +00:00
4d278e2caf Adopt absl::string_view in function parameters under audio/
This is part of a large-scale effort to adopt absl::string_view
throughout the WebRTC code base. As a first step, function parameters
of type 'const std::string&' are being converted to absl::string_view.

Bug: webrtc:13579
Change-Id: Ib4618fad3bff2902cd3a4730506aca300949d76c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252982
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36102}
2022-03-01 13:11:08 +00:00
7b0a30ec9a Allow low_bandwith_audio_test.py to pass unknown arg to the test.
* The idea is copied from flags_compatibility since this file does a bit the same thing.
* Remove extra_test_args which is not used and becomes unecessary.
* Fix lint issues.

Bug: b/197492097
Change-Id: I378e163a5116ded13619f91ce50859519c9550df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252004
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36044}
2022-02-22 09:31:47 +00:00
7a5978e4cd Revert "Add the possibility to output a json gtest output to the perf tests."
This reverts commit cbfa235b3548ae1a0b8f5b80a879214ca5f1bd56.

Reason for revert: iOS bots will use flags_compatibility with isolated_script_test_output but not gtest_output.

Original change's description:
> Add the possibility to output a json gtest output to the perf tests.
>
> We use the Chromium existing flag isolated_script_test_output that we translate into gtest_output.
> This is because the Chromium flag has the same purpose as gtest_output and is already provided in the recipe modules.
>
> No-Presubmit: True
> Bug: b/197492097
> Change-Id: Ia432a85b0e0ab32008b39ffe751d11aefb9b24ea
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251041
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#35937}

No-Presubmit: True
Bug: b/197492097
Change-Id: I94e75328570f89011fbb0daf035f0072b8ea2f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252040
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36042}
2022-02-22 07:57:34 +00:00
1db0a261ca Reland "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 09aaf6f7bcfb4da644bd86c76896a04a41f776e1.

Reason for revert: downstream fixed (see https://chromium-review.googlesource.com/c/chromium/src/+/3461371)

Original change's description:
> Revert "Reland "Remove unused APM voice activity detection sub-module""
>
> This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.
>
> Reason for revert: Breaks chromium roll, see 
> https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview
>
> https://chromium-review.googlesource.com/c/chromium/src/+/3461512
>
> Original change's description:
> > Reland "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
> >
> > Reason for revert: dependency in a downstream project removed
> >
> > Original change's description:
> > > Revert "Remove unused APM voice activity detection sub-module"
> > >
> > > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> > >
> > > Reason for revert: breaking downstream projects
> > >
> > > Original change's description:
> > > > Remove unused APM voice activity detection sub-module
> > > >
> > > > API changes:
> > > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > > - cricket::AudioOptions::typing_detection deprecated
> > > > - webrtc::StatsReport::StatsValueName::
> > > >   kStatsValueNameTypingNoiseState deprecated
> > > >
> > > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > > >
> > > > Bug: webrtc:11226,webrtc:11292
> > > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/main@{#35975}
> > >
> > > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> > >
> > > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:11226,webrtc:11292
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35977}
> >
> > # Not skipping CQ checks because this is a reland.
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35984}
>
> TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35990}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11226,webrtc:11292
Change-Id: Idfda6a517027ad323caf44c526a88468e5b52b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251762
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36012}
2022-02-16 08:41:30 +00:00
09aaf6f7bc Revert "Reland "Remove unused APM voice activity detection sub-module""
This reverts commit 54d1344d985b00d4d1580dd18057d4618c11ad1f.

Reason for revert: Breaks chromium roll, see 
https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_tsan_rel_ng/1080583/overview

https://chromium-review.googlesource.com/c/chromium/src/+/3461512

Original change's description:
> Reland "Remove unused APM voice activity detection sub-module"
>
> This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.
>
> Reason for revert: dependency in a downstream project removed
>
> Original change's description:
> > Revert "Remove unused APM voice activity detection sub-module"
> >
> > This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
> >
> > Reason for revert: breaking downstream projects
> >
> > Original change's description:
> > > Remove unused APM voice activity detection sub-module
> > >
> > > API changes:
> > > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > > - webrtc::AudioProcessingStats::voice_detected deprecated
> > > - cricket::AudioOptions::typing_detection deprecated
> > > - webrtc::StatsReport::StatsValueName::
> > >   kStatsValueNameTypingNoiseState deprecated
> > >
> > > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> > >
> > > Bug: webrtc:11226,webrtc:11292
> > > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#35975}
> >
> > TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
> >
> > Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:11226,webrtc:11292
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35977}
>
> # Not skipping CQ checks because this is a reland.
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35984}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ib308a3af2dcce85a0074ef5a4680ccec3f82712f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251688
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35990}
2022-02-14 12:25:51 +00:00
54d1344d98 Reland "Remove unused APM voice activity detection sub-module"
This reverts commit a751f167c68343f76528436defdbc61600a8d7b3.

Reason for revert: dependency in a downstream project removed

Original change's description:
> Revert "Remove unused APM voice activity detection sub-module"
>
> This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.
>
> Reason for revert: breaking downstream projects
>
> Original change's description:
> > Remove unused APM voice activity detection sub-module
> >
> > API changes:
> > - webrtc::AudioProcessing::Config::VoiceDetection removed
> > - webrtc::AudioProcessingStats::voice_detected deprecated
> > - cricket::AudioOptions::typing_detection deprecated
> > - webrtc::StatsReport::StatsValueName::
> >   kStatsValueNameTypingNoiseState deprecated
> >
> > PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
> >
> > Bug: webrtc:11226,webrtc:11292
> > Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Reviewed-by: Björn Terelius <terelius@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#35975}
>
> TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11226,webrtc:11292
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35977}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11226,webrtc:11292
Change-Id: I2fcbc5fdade16bfe6a0f0a02841a33a598d4f2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251660
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35984}
2022-02-13 14:02:08 +00:00
a751f167c6 Revert "Remove unused APM voice activity detection sub-module"
This reverts commit b4e06d032e6f82a65c52ed0c5364ae9e7c0a0215.

Reason for revert: breaking downstream projects

Original change's description:
> Remove unused APM voice activity detection sub-module
>
> API changes:
> - webrtc::AudioProcessing::Config::VoiceDetection removed
> - webrtc::AudioProcessingStats::voice_detected deprecated
> - cricket::AudioOptions::typing_detection deprecated
> - webrtc::StatsReport::StatsValueName::
>   kStatsValueNameTypingNoiseState deprecated
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0
>
> Bug: webrtc:11226,webrtc:11292
> Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35975}

TBR=gustaf@webrtc.org,saza@webrtc.org,alessiob@webrtc.org,terelius@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iee01fdb874b4e0331277f3ffe60dacaabc3859a2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11226,webrtc:11292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35977}
2022-02-11 12:15:44 +00:00
b4e06d032e Remove unused APM voice activity detection sub-module
API changes:
- webrtc::AudioProcessing::Config::VoiceDetection removed
- webrtc::AudioProcessingStats::voice_detected deprecated
- cricket::AudioOptions::typing_detection deprecated
- webrtc::StatsReport::StatsValueName::
  kStatsValueNameTypingNoiseState deprecated

PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0

Bug: webrtc:11226,webrtc:11292
Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35975}
2022-02-11 10:47:39 +00:00
cbfa235b35 Add the possibility to output a json gtest output to the perf tests.
We use the Chromium existing flag isolated_script_test_output that we translate into gtest_output.
This is because the Chromium flag has the same purpose as gtest_output and is already provided in the recipe modules.

No-Presubmit: True
Bug: b/197492097
Change-Id: Ia432a85b0e0ab32008b39ffe751d11aefb9b24ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251041
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35937}
2022-02-07 20:59:33 +00:00
4a3e56075e Switch to universal_newlines=True.
To fix the blocking issue this CL uses universal_newlines (removing the
calls to decode('utf-8')).

Tested locally.

No-Presubmit: True
Bug: webrtc:13607
Change-Id: Ib56cf87c8f903087d0c4aa09b58c464edac649c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250222
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35889}
2022-02-02 16:05:10 +00:00
d5f98ce8cf Revert Popen bufsize to Python 2.7 default value.
This might be the reason causing the process to not terminate.

This CL adds also more logging.

No-Presubmit: True
Bug: webrtc:13607
Change-Id: I9fc2cf39d2c1df92670b45dd081022ce69068836
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250181
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35885}
2022-02-02 12:55:59 +00:00
5423c83731 Switch to Popen and adding a timeout for PESQ measurements.
After migrating to python3, the check_output doesn't return, this
CL switches to communicate() with a timeout of 2 minutes (to avoid
to block bots for 2 hours in a deadlock).

No-Presubmit: True
Bug: None
Change-Id: I3248ab090c074bd35300ca11abc08536cd797664
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250164
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35882}
2022-02-02 09:46:19 +00:00
c6206658e4 Add timestamp to logs.
No-Presubmit: True
Bug: None
Change-Id: I3adc0ba9f0c92c10d35833aca2698eead40a849d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250160
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35873}
2022-02-01 18:19:08 +00:00
b6653d9967 [python3] - Fix low_bandwidth_audio_test.py (take 3)
No-Presubmit: True
Bug: webrtc:13607
Change-Id: Iff325ad10138fe8b7e1df1fa169652f5795fa718
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250081
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35865}
2022-02-01 09:11:17 +00:00
0bd9905dc4 [python3] - Fix low_bandwidth_audio_test.py (take 2)
No-Presubmit: True
Bug: webrtc:13607
Change-Id: I2cab05888d52e8964fddce233ad2903d540125fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249991
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35854}
2022-01-31 13:25:08 +00:00
b732bd5fb5 Add timestamps to AudioDeviceBuffer::SetRecordedBuffer
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.

This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.

Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
2022-01-31 12:32:58 +00:00
f3686711e9 [python3] - Fix low_bandwidth_audio_test.py
No-Presubmit: True
Bug: webrtc:13607
Change-Id: I88013e080adbafae3001cba4c1ed2428d4473d22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249984
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35839}
2022-01-31 08:37:21 +00:00
1d4e982b07 Fix python3 errors in low_bandwidth_audio_test.py.
This is causing errors on the ci:
https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Android32%20(M%20Nexus5)/3379

No-Presubmit: True
Bug: webrtc:13607
Change-Id: Ice54db8b1405623e5d873cfd2795fbf5541ef727
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249789
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35830}
2022-01-28 17:31:17 +00:00
62238097c9 Remove top-level const from parameters in function declarations.
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.

Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
2022-01-26 11:05:25 +00:00
7eac6caeee Don't use wall clock for stats
This uses the local NTP clock for RTCP report block stats.

This code exists in the version that Mozilla is shipping, with a review
here https://phabricator.services.mozilla.com/D127709 .

Bug: webrtc:13484
Change-Id: I2f46ec02acab0bbb09040778b05b248c2d815bd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240142
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35787}
2022-01-25 15:39:53 +00:00
a3361ff2f5 Update audio code to not use implicit T* --> scoped_refptr<T> conversion
Bug: webrtc:13464
Change-Id: Ida1694537b47c62ce327eb5c77897af451a63ae7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246202
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35688}
2022-01-13 15:49:49 +00:00
994bf454ec Revert of flag simplification.
In order to unify WebRTC recipes with Chromium recipes this CL tries to revert the old CL https://webrtc-review.googlesource.com/c/src/+/171681.
This CL was already partially reverted (https://webrtc-review.googlesource.com/c/src/+/171809).
In upcoming CLs, the added flag dump_json_test_results will be removed in order to use isolated-script-test-output instead.

Bug: webrtc:13556
Change-Id: I3144498b9a5cbaa56c23b3b8adbac2229ad63c37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245602
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35666}
2022-01-12 10:53:12 +00:00
7336422fe3 Delete some unneeded references to ProcessThread.
Bug: None
Change-Id: I77528df2a8bd2d461440cf59ada8229e732a1e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242370
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35613}
2022-01-03 15:36:02 +00:00
f22c78b01a Fix mb.py presubmit issues.
* Add a config file for python formatting (.style.yapf).
* Change the default indentation from 4 spaces to 2 spaces.
* Run 'git cl format --python' on a few python files.

Bug: webrtc:13413
Change-Id: Ia71135131276c2c499b00032d57ad16ee5200a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238982
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#35500}
2021-12-08 08:53:00 +00:00
a2e3d80cf6 Revert "Reland "Use gtest_parallel with 1 worker for webrtc_perf_tests.""
This reverts commit c31fc2a941d417286e1a56d6f18d5051c99d06d9.

Reason for revert: Fix is not working properly.

Original change's description:
> Reland "Use gtest_parallel with 1 worker for webrtc_perf_tests."
>
> This is a reland of 258ed1a38ad9d4f0da798c40b6976eff2dce864f
>
> Original change's description:
> > Use gtest_parallel with 1 worker for webrtc_perf_tests.
> >
> > This will enable test results to be uploaded to ResultDB.
> >
> > Bug: b/197492097
> > Change-Id: Iec28520c4cd8f35fcff2cbd105a4b851ef41b9fc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239641
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Christoffer Jansson <jansson@google.com>
> > Commit-Queue: Jeremy Leconte <jleconte@google.com>
> > Cr-Commit-Position: refs/heads/main@{#35458}
>
> Bug: b/197492097
> No-Presubmit: True
> Change-Id: Iea90f5698c83791d39c0f6da666c1d1eb274edd3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239645
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35483}

TBR=mbonadei@webrtc.org,gustaf@webrtc.org,jansson@google.com,jansson@webrtc.org,jakobi@webrtc.org,landrey@webrtc.org,jleconte@google.com,jleconte@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Iee9b67db99545a1e6c707bc03faaf55afc90cbbf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/197492097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240182
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35486}
2021-12-07 12:04:08 +00:00
c31fc2a941 Reland "Use gtest_parallel with 1 worker for webrtc_perf_tests."
This is a reland of 258ed1a38ad9d4f0da798c40b6976eff2dce864f

Original change's description:
> Use gtest_parallel with 1 worker for webrtc_perf_tests.
>
> This will enable test results to be uploaded to ResultDB.
>
> Bug: b/197492097
> Change-Id: Iec28520c4cd8f35fcff2cbd105a4b851ef41b9fc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239641
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Jansson <jansson@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#35458}

Bug: b/197492097
No-Presubmit: True
Change-Id: Iea90f5698c83791d39c0f6da666c1d1eb274edd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239645
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35483}
2021-12-07 10:08:16 +00:00
f47a724168 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}
2021-11-23 08:37:36 +00:00
d3251968d1 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}
2021-11-15 21:44:59 +00:00
bf0874568c Implement RTCOutboundRtpStreamStats.targetBitrate for audio.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate

Bug: webrtc:13377
Change-Id: I98dd263e0b9d6e2ca94969d2a91857b14cd65f70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237402
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35337}
2021-11-12 09:24:34 +00:00
1d73243466 Use a new instance of RTP stack for each test.
- Reusing RTP stack may have contributed to some flakiness as
  the previous state could have persisted to new test being performed.

Bug: webrtc:13241
Change-Id: Idf70b56bd3377bc99321fddf7191d7a72c37b085
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237540
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35336}
2021-11-11 17:33:03 +00:00
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
2021-11-04 13:46:17 +00:00
bd9106d88f voice_engine: dont announce rid/rrid header extensions
which do not make sense for audio due to lack of support for RTX.

BUG=webrtc:13279

Change-Id: Ida42d8912bf993f01e0dc5c6ffbdbf4b84495c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235061
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35309}
2021-11-04 12:47:48 +00:00
723b35f6f0 Delete legacy function to deregister rtp header extension by type
Bug: None
Change-Id: I1d9447df41edf109665a5c746a6dc2c912aad8a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234526
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35179}
2021-10-11 15:42:19 +00:00
8fb41a39e1 Add Direction indicator to TransformableFrames
Currently the implementation of FrameTransformers uses distinct,
incompatible types for recevied vs about-to-be-sent frames. This adds a
flag in the interface so we can at least check that we are being given
the correct type. crbug.com/1250638 tracks removing the need for this.

Chrome will be updated after this to check the direction flag and provide
a javascript error if the wrong type of frame is written into the
encoded insertable streams writable stream, rather than crashing.

Bug: chromium:1247260
Change-Id: I9cbb66962ea0718ed47c5e5dba19a8ff9635b0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <toprice@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35100}
2021-09-27 17:40:36 +00:00
d0321c5e5a Deduplicate set of the rtp header extension uri constants
Bug: webrtc:7472
Change-Id: Ic0b4f2cc3374ba70a043310b5046d8bf91f0acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231949
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34990}
2021-09-14 13:38:44 +00:00
2562cf0105 Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb.

Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.

Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
2021-09-06 14:26:55 +00:00
2c41cbae37 Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.

Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.

Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}

# Not skipping CQ checks because original CL landed > 1 day ago.

TBR=hta,hbos,minyue

Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
2021-09-01 17:32:00 +00:00
9adbbebde9 Removing usage of std::unordered_set
In Chromium, they are discouraged.

Bug: webrtc:12689
Change-Id: I0e2a03b909d8a6d239e11969659e4fdc1a89766c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229188
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34880}
2021-08-31 08:59:41 +00:00
fb0dca6c05 Wire up non-sender RTT for audio, and implement related standardized stats.
The implemented stats are:
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements

Bug: webrtc:12951, webrtc:12714
Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34861}
2021-08-30 09:03:50 +00:00
2ace42f084 frame transformer: expose payload type
spec PR: https://github.com/w3c/webrtc-encoded-transform/pull/117

Bug: webrtc:13077
Change-Id: I81d79201cea353c26ea840e92c0deec7c7253b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34844}
2021-08-25 08:33:20 +00:00
81f24c09fe Add jakobi@webrtc.org to audio/OWNERS
Also removing saza@webrtc.org and sorting the list

Bug: None
Change-Id: I24fd0d9c82af1bb61bcfb5b3987504e066d05ef7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229305
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34807}
2021-08-19 15:30:32 +00:00
ac09f0dc92 Remove last traces of deferred sequencing.
Bug: webrtc:11340
Change-Id: I761be67d42959192355f9f6f54ed1f735da1fe96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228646
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34770}
2021-08-16 12:44:37 +00:00
69dd142797 Register audio send stream in packet router on Start().
Currently, the RtpRtcp module of AudioSendStream is (de)registered in
the packet router on calls to
(Register|Reset)SenderCongestionControlObjects.
This CL changes that to happen on Start/Stop instead, which allows us
to safely call (Get|Set)RtpState on suspend/resume without the need
for extra locking in the rtp module.

See also https://webrtc-review.googlesource.com/c/src/+/228430

Bug: webrtc:11340
Change-Id: I54243a9ace8a7659924269418468b49b967b9465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228433
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34738}
2021-08-12 15:15:53 +00:00
6144b8422b red: fix renegotiation
If RED is no longer used the send codec needs to be reconfigured.
To test on https://webrtc.github.io/samples/src/content/peerconnection/audio/
run:
  await pc1.setLocalDescription();
  await pc1.setRemoteDescription({type: 'answer', sdp:
        pc1.remoteDescription.sdp.replace('red/48000', 'blue/48000')})
As a result, RED will be turned off and the bitrate will drop.

BUG=webrtc:11640

Change-Id: Icc7a83ae29e67d054399bf42010264e94c32127d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221360
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34719}
2021-08-11 12:00:13 +00:00
6c02c33df9 Add henrik.lundin as owner in audio/
Bug: none
No-Try: True
Change-Id: I9de4fab3b1db29c91396c371395d9d3399c80239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228427
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34714}
2021-08-11 08:45:59 +00:00