Lifetime issue: "webrtc_audio_module_rec_thread" was still accessing
AudioTransport mock at and after its destruction.
Bug: webrtc:9751
Change-Id: I24308077cdeb77e570b8ec74098f1ae3397b7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146217
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28635}
Emscripten does not support C++11 thread_local but does support
the pthread TLS API.
Bug: None
Change-Id: Ia21895148d1df7652579d086d9e1c0c53d7a85f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145441
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28621}
This tool is unused, this CL removes it in order to reduce the cost
of the maintenance (in the last 2 years only maintenance commits have
been landed in this directory).
Bug: None
Change-Id: Ieec113bc25c480405d32e284a0456572758352e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146204
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28619}
Rationale:
* More explicit (you won't miss that when glancing at the code).
* More consistent (see MAYBE_* in other tests).
* Allow to re-activate tests via CLI (--gtest_also_run_disabled_tests).
* Tests won't wrongly show up as PASSING (bug/webrtc:10819),
since they won't show up at all.
Bug: webrtc:9778
Change-Id: Ic32e18cb8ee2352def95206c2aa66e1dea0cc1e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146200
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28617}
This CL removes the field trial left in place as a kill-switch in case
there were any regressions related to selecting payload padding based
on the likelihood of being useful instead of matching size.
It also removes the functionality that was only enabled with the
kill-switch active.
The feature has been default-on since June 23rd 2019:
https://webrtc.googlesource.com/src.git/+/214f54365ec210db76218a35ead66c9ce23e068e
Since we have not observed any issues, let's clean this code up.
Bug: webrtc:8975
Change-Id: I7f49fe354227b3f6566a250332e56b6d70fe2f09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145821
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28616}
This CL fixes two things related to the (not yet active) new
PacedSender code path:
1. Make sure BWE header extensions are properly populated for all
padding packets.
2. When generating padding, don't hold the RtpSender critsect when
accessing the RtpPacketHistory as this may lead to a lock order
inversion.
Bug: webrtc:10633
Change-Id: I8650fbf5dafddbeae61837d2137338163e1c48ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145723
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28613}
SVC support is limited:
During SVC testing there is no SFU, so framework will try to emulate SFU
behavior in regular p2p call. Because of it there are such limitations:
* if |target_spatial_index| is not equal to the highest spatial layer
then no packet/frame drops are allowed.
If there will be any drops, that will affect requested layer, then
WebRTC SVC implementation will continue decoding only the highest
available layer and won't restore lower layers, so analyzer won't
receive required data which will cause wrong results or test failures.
Bug: webrtc:10138
Change-Id: I079566260ca9f1815935bce365d1bca10766663a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144882
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28612}
This API is going away, we'll use the WebRTC-Audio-Allocation field
trial flag to set this value in the future.
Bug: webrtc:10556
Change-Id: I2c4c1948a33f909fac069dd038cea36a793e4745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145405
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28608}
This is the last CL required to migrate WebRTC to ABSL_FLAG, rtc::Flag
will be removed soon after this one lands.
Bug: webrtc:10616
Change-Id: I2807cec39e28a2737d2c49e2dc23f2a6f98d08f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145727
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28606}
Landing with TBR given vacation times and the fact that none of this
code is active "in production". The ADM2 implementation can be seen
as experimental (non-default) code and it takes some work to enable it
and replace the existing ADM. Hence, extremely low risk to break
anything.
TBR: henrik.lundin
Bug: webrtc:9265
Change-Id: Ibc9a57f4851bf4b890b77b9eaef1dfbe3ca86f83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146084
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28601}
Landing with TBR given vacation times and the fact that none of this
code is active "in production". The ADM2 implementation can be seen
as experimental (non-default) code and it takes some work to enable it
and replace the existing ADM. Hence, extremely low risk to break
anything.
TBR: henrik.lundin
Bug: webrtc:9265
Change-Id: Ia5cfb2aaa8eaf9537b916b3375f55d8df6287071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145921
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28600}
The simulcast allocator would only set bitrates for the first 2 layers
in conference_screenshare_mode.
That would trigger an issue in the VP8 encoder initialization that expects
to have growing bitrates for the layers (3rd layer would have the same
bitrate as the 2nd one).
Bug: webrtc:8785
Change-Id: Ic6c940b78022387841b28074b373be6b2f45cb15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145922
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28598}
Marking test as skipped is more honest than pretending it is successful!
Prevent confusion like in the following scenario for one given test:
- ubsan: launched and sometimes failing.
- tsan: never launched but always flagged OK.
Bug: webrtc:9778
Change-Id: Ie0be0759347eabd3c9d29dd5ea2de809511d1b97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145980
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28597}
This is a temporary solution, as there are several other executables and
some tests in rtc_tools/BUILD.gn. Including all of them to default target
is not decided yet.
But as rtp_generator tends to be broken reguraly, It should be included
there at least for now.
Bug: webrtc:10807
Change-Id: I3acf5a93c74bf1e2474c6aaee35653efbb43d3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28595}
This is a reland of bd33ce26202272177af6c52e195e7c13f0d1bf23
Now it doesn't apply flags_compatibility to Android, because the device runner actually requires it to be dashes (so it can intercept the flag and substitute it with an Android-local file path), but that's OK because the runner also already passes the flag with underscores: https://cs.chromium.org/?q=%22--isolated_script_test_perf_output%22
Original change's description:
> Reland "Add wrapper to normalize flags."
>
> This is a reland of 642a49d1eb20b8c5744e745de79ddb585e0f7472
>
> The change has the same effect but is now implemented through mb, rather than specifying a 'script', so that Android's special handling is not skipped.
>
> Original change's description:
> > Add wrapper to normalize flags.
> >
> > Bug: None
> > Change-Id: I9d43602cc66198a29dbc0e7586d948ee76c5ec84
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145204
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28532}
>
> Bug: webrtc:10616
> Change-Id: I60ebd4891dbe8de18c653f8af88181ea966307de
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145409
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28572}
Bug: webrtc:10616
Change-Id: I56aae5475aed62f069c5cecc01b75d7d6ffcf568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145920
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28589}
