To mitigate race between ~ChannelSend and task created in ProcessAndEncodeAudio.
as described in the comment next to the task queue member.
Bug: b/228933184
Change-Id: Ia0efd050c76a4539dc2525ef8efc065fab96861c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258983
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36553}
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}
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}
* 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}
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}
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}
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}
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}
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}
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}
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}
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}
* 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}
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}
- 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}
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}
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}
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}
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}
