It is possible to have AddTransceiver calls with an empty array
of encodings or AddTrack calls. In both cases, before negotiation,
the sender's encodings array would be empty and it was not possible
to update any value.
Now, a default entry should be created in those cases, allowing to
update the configuration before negotiation.
Bug: webrtc:10567
Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39126}
Remove the default enabled "WebRTC-Audio-FixTimestampStall" field trial which was rolled out 2 years ago without any issues.
Also change the include audio level indication member to be atomic since it is accessed on multiple threads.
Bug: webrtc:14804
Change-Id: I4c5145e1fb03351154162b4293a3bd870e4793cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290886
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39125}
This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate.
Bug: webrtc:14818
Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39123}
optimizing for the fairly common case of many recv-only
mediasections.
BUG=webrtc:14808
Change-Id: Iae68c5bb7a5516d77f908f1effbb50a5ed750f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290984
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39122}
Implement GetProcessCpuTimeNanos and GetThreadCpuTimeNanos for Fuchsia.
This is needed for the tests call_perf_tests and
video_pc_full_stack_tests on Fuchsia.
Bug: fuchsia:115601
Change-Id: Idd10db93d4087d10896ae3fde6abbc37176f625e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290920
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sarah Pham <smpham@google.com>
Cr-Commit-Position: refs/heads/main@{#39119}
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.
This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
disabled and TS is enabled
2. when the initial APM sample rate is different from the
capture one and the VAD APM sub-module is not re-initialized
This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.
Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
instead of the full set of codecs that have been negotiated.
BUG=webrtc:14808
Change-Id: I464cc1d20e5b5227a09929c909615b432c6be041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39114}
RTPVideoHeader is changed to non-const to allow modifying it. We want
to do this when implementing setMetadata() in JavaScript or when
refactoring clone() as "construct + set bytes + setMetadata".
Unblocks
https://chromium-review.googlesource.com/c/chromium/src/+/4164979.
Bug: webrtc:14709
Change-Id: I6089df9c03e9aa33feeb0830dd240dd456cb565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39113}
To be able to simulate offline some scenario in which the javascript
layer set the minimum base buffer size of neteq, it is required to
record those API calls. This change introduces this.
Bug: webrtc:14763
Change-Id: Ic817913eda60978d6fca3f8e12229aeec505ca25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287122
Auto-Submit: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39104}
This is now ready for plumbing to Chromium layers.
Once it's exposed in JavaScript (behind flag!) we can evaluate whether
all of this information is really needed or if the information is
superflous (e.g. already contained in the raw bytes).
Bug: webrtc:14709
Change-Id: I3837ef86046704a300ec8a108c8c9477bd91b9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290884
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39102}
This will allow exposing VP8, VP9 and H264-specific RTP header metadata
in JavaScript (behind a flag).
This information appears to be necessary for cloning
(https://github.com/w3c/webrtc-encoded-transform/issues/161), and
cloning should be the same as "new frame + setMetadata + setBytes",
ergo this should be exposed.
Bug: webrtc:14709
Change-Id: Ie71c05f40689bbd529dc4674a07a87c7910b22d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290880
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39101}
This is needed for Chromium. The video capture API in Chromium expects the
raw frames and it will always convert or copy the frame. With the existing
API that would mean copying the frame twice.
Bug: webrtc:13177
Change-Id: I71f6e2dc6d5a812c3641ac691b75d50178fa0de7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264548
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39095}
Also set xcode back to xcode 13 for iOS 14.
Change-Id: Ic5475d274895b5f86e4fea36805dec4486adc79b
Bug: b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290894
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39094}
The goal of this bot is to replace ios_sim_x64_dbg_ios(12, 13 and 14).
Change-Id: I6d8f5004a9440f5fd8cb96730dc2dbb4abba2e61
Bug: b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290893
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39086}
There is now only one sequence number unwrapper so this is redundant.
Bug: webrtc:13982
Change-Id: I210378e069366eb21dea0051f83d7431e4177057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290892
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39085}
PacketArrivalMap explicitly doesn't promise packet at the beginning
of it is received. Ensuring that property is wasteful
Bug: chromium:1382563
Change-Id: Ifc898b7ec2bc7a302af8dcfd233e0c598f62db95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290501
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39083}
