We need to keep the lock until we have finished using the task runner
returned by GetNextReadyRunner to ensure that we don't remove it from
another thread. Additionally, we must get only one runner at a time in
case the first runner removes the second runner.
Bug: webrtc:10538
Change-Id: Idbd5610b67666238545b3a321fb79f7e86fcac56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132342
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27584}
This reverts commit 7ac0d5f348f0b956089c4ed65c46e65bac125508.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This CL just moves code around to prepare for an upcoming
CL where more stats collection is added to scenario tests.
Bug: webrtc:10365
Change-Id: I8a960e44fd11fc36047677c4d8dfc0af96aacb22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132002
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27519}
This CL does two things:
* It ensures that video quality analysis is always finished when a
Scenario test is stopped. This ensures consistency between different
task queue implementations.
* It disables one real time test that is too heavy to run on IOS
ARM64 debug builds.
Bug: webrtc:10515
Change-Id: I34b59ecde6f2b68c399734a43ecdbc7223725b17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131388
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@google.com>
Cr-Commit-Position: refs/heads/master@{#27495}
See https://webrtc-review.googlesource.com/c/src/+/121764 for the
overall vision.
This CL adds a multistream Opus decoder. It's a new code-path to not
interfere with the standard Opus decoder. We introduce new SDP syntax,
which uses terminology of RFC 7845. We also set up the decoder side to
parse it. The encoder part will come in a later CL.
E.g. this is the new SDP syntax for 6.1 surround sound:
"multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2"
Bug: webrtc:8649
Change-Id: Ifbc584cbb6d07aed373f223512a20d6d72cec5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129768
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27493}
rtcp_receiver_fuzzer was running over inputs of unreasonable
length, leading to timeouts. RTCP typically runs over UDP.
This CL limits the inputs to a bit over the max UDP payload length.
Bug: chromium:948469
Change-Id: I669a5b24c265bb3b6da2503da109efed32c25182
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131393
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27482}
The DefaultAudioQualityAnalyzer will read stats reports (temporarily
using the old PeerConnectionInterface::GetStats) and for each audio
stream it will collect some NetEq related stats.
When DefaultAudioQualityAnalyzer::Stop is invoked by the framework,
it will report the following metrics:
- expand_rate
- accelerate_rate
- preemptive_rate
- speech_expand_rate
- preferred_buffer_size_ms
Bug: webrtc:10138
Change-Id: Ie493456fcb9ed86455b12dabdab98a317387ef46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125980
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27474}
This is to avoid inconsistent/flaky behavior on mobile bots.
Bug: webrtc:10365
Change-Id: I52ab4f9ef92b10329c1eac502adfcf2886058114
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27465}
Refactoring of quality measurement code, basing frame matching on
frame thumb likeness. This way the code is robust against variations
in timing and frame drops.
Bug: webrtc:9510
Change-Id: Ief7266e01f39ca621a529c0da736e5ed1df8560a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27415}
The has some benefits:
* We no longer need locks to protect the emulated network node state.
* We only process when there are packets in flight.
* It makes Scenario more similar to network emulation manager.
Bug: webrtc:10365
Change-Id: I8bd1ad1edfb54b047e8109dabd9846ae451cef17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127548
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27393}
As part of this change, a task queue is used to handle packet
processing in real time mode. This requires that we also do
most call and media stream related operation on the same task
queue to satisfy thread checkers.
Bug: webrtc:10365
Change-Id: Icdd9d56e4ca14f2c944dc655c91e29392e3765f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127544
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27379}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
It is possible for the fuzzer to just never deliver packets if the packet delay
is set long enough in the RtpReplayer. This is simply fixed by setting an upper
bound. This change is in the test code setup.
Bug: webrtc:10493,chromium:943420
Change-Id: I54f56e1aa7700f1151e0b58a5a53bc789d032c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130365
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27369}
Replacing sets of pointers (that will depend on allocation addresses)
with vectors and lists. This allows deterministic execution.
Also doing some cleanup of the task queue configuration, ensuring that
the task queue states is not set outside of actual task queues.
Bug: webrtc:10365
Change-Id: I1fad621c7b1ba0bbb33db8c3bd69cb3a1e212b9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27364}
