The lock is unnecessary and potentially unsafe:
1) All gain_control accesses in AudioProcessingImpl happen - and are intended to happen - while holding the crit_capture_ lock, and all external API calls take the same lock once inside GainControlImpl.
2) If ProcessCaptureStreamLocked (locked by crit_capture) calls a gain_control function that takes crit_render, the mandated locking order (render before capture) is violated and we might get a deadlock with the render thread.
Bug: b/123456404
Change-Id: Id7a888827e347e5e1d50e2f87d90e8b68f52b7b8
Reviewed-on: https://webrtc-review.googlesource.com/c/122087
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26637}
This reverts commit 5054f544575b1a0471b241266c6fc8c2ccf93af0.
Reason for revert: Partial Capture API is not needed, according to new info from the Chrome team.
Original change's description:
> Partial frame capture API part 2
>
> Implement test utility for extracting changed part of video frames.
>
> Bug: webrtc:10152
> Change-Id: Iead052d2a18384aaa828cd7821be961b8614568e
> Reviewed-on: https://webrtc-review.googlesource.com/c/120407
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26496}
TBR=ilnik@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10152
Change-Id: I80cae8a7d352b4ee67b42f5388fd8c1883ab2e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/122091
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26632}
Currently there's an implicit requirement that users of
SimulatedNetwork should call it repeatedly, even if the return value
of NextDeliveryTimeUs is unset.
With this change, it will indicate that there might be a delivery in
5 ms at any time there are packets in queue. Which results in unchanged
behavior compared to current usage but allows new users to expect
robust behavior.
Bug: webrtc:9510
Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069
Reviewed-on: https://webrtc-review.googlesource.com/c/120402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26617}
with the value that actually ends up being assigned here. There is no change in actual behavior.
Bug: None
Change-Id: I268c50a920a5d7e98909a9ec760fc80ca0718417
Reviewed-on: https://webrtc-review.googlesource.com/c/121540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26565}
This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.
It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump
It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.
Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
It doesn't do anything any more, so it should be removed.
Bug: webrtc:9586
Change-Id: I0b320b6ce4f480ff8cb59451db29bcc77b882b5f
Reviewed-on: https://webrtc-review.googlesource.com/c/120813
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26507}
Implement test utility for extracting changed part of video frames.
Bug: webrtc:10152
Change-Id: Iead052d2a18384aaa828cd7821be961b8614568e
Reviewed-on: https://webrtc-review.googlesource.com/c/120407
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26496}
Create audio stream instead of data channel to check compatibility of
network layer with PeerConnection. Replacement is done because there is
a data race inside data channel sctp transport. This CL will fix
bot behavior. Further data race investigation will be done in this
bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=10268
Bug: webrtc:10268, webrtc:10138
Change-Id: I4f7a1116c65dbf4a3508b7d81d654ccd320795f0
Reviewed-on: https://webrtc-review.googlesource.com/c/120807
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26495}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:10270
Change-Id: I316f9788adac954c52b0f9230881b872c54a7ac9
Reviewed-on: https://webrtc-review.googlesource.com/c/120348
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26482}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
TestPeer represent single participant in the call and will own most
required for call objects.
TestPeer::CreateTestPeer is responsible for full setup of TestPeer and
allow to correctly inject media analyzers into call.
Bug: webrtc:10138
Change-Id: Ide7062004b0dc113b9c05181d8144797a3cc27a8
Reviewed-on: https://webrtc-review.googlesource.com/c/119941
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26464}
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)
Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
Add static factory method from FrameGenerator for FrameGeneratorCapturer
to be able to intercept generated frames in PC e2e test framework to
dump input video stream into file, if it was generated.
Bug: webrtc:10138
Change-Id: Iabecfaaef804111e0b19756cd676c1749757d9c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119947
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26424}
It didn't have proper time to stabilize, making it sensitive to small
changes. This CL increases the stabilization period from 20 to 30s.
Also fixing some minor test suite bug found during investigation.
Bug: webrtc:9718
Change-Id: If56dba5383251ad3d3efe304eebcd880522afabe
Reviewed-on: https://webrtc-review.googlesource.com/c/119943
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26408}
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
VideoQualityAnalyzerInjectionHelper will be used to provide all required
entities to inject video quality analyzer into peer connection pipeline.
Bug: webrtc:10138
Change-Id: Iea7cf453311d809619839d5cf94b78a020ce9167
Reviewed-on: https://webrtc-review.googlesource.com/c/119642
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26395}
This encoder will be used to inject VideoQualityAnalyzerInterface into
VideoEncoder, so it will be able to measure its metrics and also trace
frames from capturing on one peer side to rendering on another peer side.
The decoder will be used for the same purpose but in VideoDecoder pert.
Bug: webrtc:10138
Change-Id: Idf719753e3c0b3b1369ff206365bf0558705eb98
Reviewed-on: https://webrtc-review.googlesource.com/c/117363
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26381}
The FrameGeneratorCapturer instances continue to live after
RunBaseTest() returns, and have their own internal task queues. This
means any class that listens for frames may be called after return
from RunBaseTest(), at which point they may be destroyed.
This CL makes sure we remove any capturer before returning.
A specific example of this problem is
VideoSendStreamTest.SuspendBelowMinBitrate
Bug: None
Change-Id: I857566301acce3e32c0888c7a1d2ee3470e6eb28
Reviewed-on: https://webrtc-review.googlesource.com/c/116684
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26375}
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.
Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
In particular, time_utils.h is currently pulled in via rtc_event.h
This CL is in preparation of moving parts of the RTC event log to api/.
Bug: webrtc:10206
Change-Id: Idd35aa9404afded4d29b1296344996c45b8c2e91
Reviewed-on: https://webrtc-review.googlesource.com/c/117921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26326}
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.
It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.
Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
VideoFrameWriter is designed to accept webrtc::VideoFrame as input and
write it with Y4mFrameWriterImpl to the output file, transforming
webrtc::VideoFrame to the uint8_t* frame_buffer. VideoFrameWriter will
be used to write webrtc::VideoFrames during dumping input and output
video in peer connection level test framework and will be injected
in webrtc::test::FrameGenerator and rtc::VideoSinkInterface<VideoFrame>.
Bug: webrtc:10138
Change-Id: Iadec7d3ad66f226836acbebe070cf88ceb242f62
Reviewed-on: https://webrtc-review.googlesource.com/c/117200
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26305}
This analyzer will be used in implementatino of peer connection level
test framework before main analyzer will be implemented.
Bug: webrtc:10138
Change-Id: Ibb7c5cd94b0f07c6fc5a2415f04b0f0ae7ae75e2
Reviewed-on: https://webrtc-review.googlesource.com/c/117221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26301}
