This reverts commit 0e2221eb2f02ed950f4fd9c7fea40b382ea0a0c8.
Reason for revert: Speculative revert, breaks downstream.
Original change's description:
> Use ADM internal state for init state check.
>
> When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.
>
> Bug: webrtc:14054
> Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Tim Na <natim@google.com>
> Commit-Queue: Tim Na <natim@google.com>
> Cr-Commit-Position: refs/heads/main@{#36893}
Bug: webrtc:14054
Change-Id: I1fa0a1ff440b9619aba60ec25970ce88a67739db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262660
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36896}
When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.
Bug: webrtc:14054
Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@google.com>
Commit-Queue: Tim Na <natim@google.com>
Cr-Commit-Position: refs/heads/main@{#36893}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
This reduces the surface of externally accessible state that belongs
to the class, which makes it easier to control what state belongs to
what thread. In this CL enforcing remote_ssrc() to be conceptually const
and sync_group to conceptually belong to the packet delivery thread.
Bug: webrtc:11993
Change-Id: I7de9366dc0c2bf451b5c58595c2d073b4016f2e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261450
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36813}
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}
