native_test_jni_onload depends on base_jni which depends on modules/audio_processing:api. This requires to include audio_device_java in pure video targets like video_codec_perf_tests.
Bug: webrtc:14852
Change-Id: I5e7b102fd730801562695bf3f4d5170ec8e59b58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301363
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39873}
MaybeWorkerThread* GetWorkerQueue() and is removed.
Instead all work is expected to be done on the taskqueue used when
creating the RtpTransportControllerSend.
Bug: webrtc:14502
Change-Id: Iedc30efb8de7592611d6d3c5b5c6cd33c17a60c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300867
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39872}
This was found to be uninitialized by an internal MSAN bot.
Bug: b/276318905
Change-Id: I0f0742113b6a5eba10ec6f51072510c91bf5676b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301401
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39871}
Ensure probes are not created until after the transport becomes writable even if the network route change.
DTLS negotiation must complete before there is a point in sending probes.
Bug: webrtc:14928
Change-Id: Ib3cb93aef9f38e306b094dd700ed595cf9eb3f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301362
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39870}
This is what Firefox implementation relies on and I can see that also
the V4L2 implementation is doing the same.
Bug: webrtc:15087
Change-Id: I641062ba879b6ef83e31af79ecc9d06fdae54adb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301320
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39869}
The jitterbuffer would call Flush which takes a mutex from
InsertPacket, which is already under the same mutex. Fix
this by introducing an internal flush method that assumes
a locked state.
The change also adds more thread annotations in case more
problems were present. No more problems were detected.
Fixed: b/277930190
Change-Id: If85609f27f8187ade9370847fecc2bc83d940dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301340
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39868}
This reverts commit 2ec6a6c57830e06f601607c1b9473ad821b57e07.
Reason for revert: It breaks WPT tests (e.g. https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1361972/overview) blocking the roll into Chromium.
Original change's description:
> Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code.
>
> * DCC = DataChannelController.
>
> * Consolidate steps to set the mid and transport name. They're now
> set at the same time and without a separate PostTask.
> * Transport sink is now consistently set in DCC
> * Order of notifications for setting up the transport is now the same
> regardless of the first time the transport is being set or if it's
> being replaced.
> * Made set_data_channel_transport() private.
>
> Bug: webrtc:11547
> Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39859}
Bug: webrtc:11547
Change-Id: I0d8d7453b71be80fbf1b7eba7d161336e29de091
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301360
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39864}
This creates the RtpCodec structure for the common fields
used in RtpCodecParameters and RtpCodecCapability.
Remove the unused fields from both that were defined from ORTC
and never implemented as well.
Bug: webrtc:15064
Change-Id: I37b4c83e2051a888fc99cc0d9f7aeb8d74f0421d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301182
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39862}
* Removes a ~60Hz thread-wakeup signal caused by the FrameArrived event
* Initial power measurements shows a reduced power consumption
* No increase in time to first captured packet found
Bug: chromium:1428592
Change-Id: Ia23b5db0c87e70e5b0d6919394494a24d8944493
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301200
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39861}
* DCC = DataChannelController.
* Consolidate steps to set the mid and transport name. They're now
set at the same time and without a separate PostTask.
* Transport sink is now consistently set in DCC
* Order of notifications for setting up the transport is now the same
regardless of the first time the transport is being set or if it's
being replaced.
* Made set_data_channel_transport() private.
Bug: webrtc:11547
Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39859}
This CL removes the usage of the Magnifier screen capture API on
Windows. The idea is to remove the actual source in a second step
once this change lands.
Bug: chromium:1428341
Change-Id: Id2cb25632c7edbea2cf527959b14b27ee00b0e56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301164
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39856}
These tests often fail in 'ExtrapolateLocalTime' because the result gives a negative Timestamp.
Here is the stack from https://chromium-swarm.appspot.com/task?id=6173230e67897b10:
PC: @ 0x7f03afdb8e87 (unknown) raise
...
@ 0x55f4a360ba71 352 webrtc::Timestamp::operator+()
@ 0x55f4a47ecaf3 160 webrtc::TimestampExtrapolator::ExtrapolateLocalTime()
Low-Coverage-Reason: coverage isn't that low.
Change-Id: If3e7cbf31d6c4800727b24352ed2c6edc425fc73
Bug: webrtc:15022
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39853}
More or less bit-exact, only difference is that we don't seek in the
input file before returning silence for DTX packets.
Bug: webrtc:13322
Change-Id: I147b70d4a0f2c78719c9673b55df6617e064bd61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301104
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39851}
This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.
Also it will allow to remove WaitForRecordingEnd() method from Test
ADM
Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}
Since PacketRouter now is only used on the worker thread, there is no need for a lock.
Bug: webrtc:14502
Change-Id: I65778f68b7e211d7bc7388a4615888a49ceb5f59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300964
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39848}
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.
This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.
Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
This consolidates termination logic in the DataChannelController
to make shut down consistent between when the transport notifies
of termination and when termination is initiated from the PC side.
This removes the need for `OnTransportChannelClosed` from the PC
side since we can just use TeardownDataChannelTransport_n (the two
were always being called together).
Bug: webrtc:11547
Change-Id: I1763f82cbfe1a3d5b8bfabb8d4cba0ee0fa95738
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300561
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39838}
The idea is to land this in Canary and ask for feedback from users
who can reproduce the issue, solve the issue and then revert this CL.
Example: https://paste.googleplex.com/6080504230051840
Bug: chromium:1421656
Change-Id: Ic214dc341a322470970abeca1794493f45b93843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301080
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39834}
This adds the histogram WebRTC.Audio.AudioMixer.NewHighestSourceCount
which logs the highest number of sources an AudioMixer has had. The
statistic is logged whenever the highest number of sources increases.
This allows us to differentiate the statistic to see how many times
the mixer has had a certain maximum number of sources.
Chromium CL:
https://chromium-review.googlesource.com/c/chromium/src/+/4414896
Bug: chromium:1430806
Change-Id: Iab92e201a0b667741cc8f3bbbed92fa989d7fcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300860
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39833}
Applying thread guards and removing the accessor that was being
called from the wrong context.
Bug: webrtc:11547, webrtc:9987
Change-Id: I80953aab48e5d155fc9d101526a3fa1f2704c39f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300544
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39832}
Bug: chromium:1431897
Change-Id: Ib871dc22d2cf93180d7aa05016e34ffec944d73e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301040
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39830}
This component is mostly "glue" and is heavily tested in the
socket tests, but not the ToString method, which results in
getting "low test coverage" warnings.
So for the sake of it, add a test that verifies that it works.
Bug: None
Change-Id: Id2b75e2798f334452be50631ef1ff15f53fe4157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300441
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39826}
