This reverts commit fc5d627cef71f906e921476c2e6b1e01d07732fe.
Reason for revert: Breaks upstream WPT tests
Original change's description:
> Reland "Ensure RTCRtpSenders are always created with one encoding"
>
> This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
>
> Original change's description:
> > Ensure RTCRtpSenders are always created with one encoding
> >
> > 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}
>
> Bug: webrtc:10567
> Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39145}
Bug: webrtc:10567
Change-Id: If9b5adb5debb7c87a15662a8d0f232405a0e8136
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291221
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39147}
This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
Original change's description:
> Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
>
> PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> Therefore DirectTransport is provided with the extension mapping.
>
> CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
>
>
> Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> Bug: webrtc:7135, webrtc:14795
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39137}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39146}
This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> 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}
Bug: webrtc:10567
Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39145}
This is used by CRD and export is required for component builds
to work properly.
Bug: chromium:1291247
Change-Id: I281e490b7d00cbd074b96eac905976af38400f8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291200
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39143}
PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
Therefore DirectTransport is provided with the extension mapping.
CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
Bug: webrtc:7135, webrtc:14795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39137}
Receivers no longer need to set extensions in the configuration. That field will be removed in a follow up.
Tested with:
video_loopback --rtp_dump_name="./my.rtpdump" --duration=10
video_replay --input_file=./my.rtpdump
Bug: webrtc:14795
Change-Id: I952cd487cb2f3be8be01a90f6a2312f1fef5d93e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290995
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39136}
This reverts commit b8023690d9f0e150cfe698cd68b477903ac66178.
Reason for revert: Breaking WPT tests in Chrome. Example build https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1263191/overview
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> 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}
Bug: webrtc:10567
Change-Id: Ib8931b38182251baac616540788a02a5fafaf670
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291120
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39132}
This CL removes the iOS sim bots 12, 13, 14 and replaces them with a unique bot that runs the test on iOS version 14, 15 and 16.
Change-Id: I46673bad44c2c5539fbbf266cc9d5d468557022e
Bug: b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290999
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39131}
The test is already disabled on iOS < 16 and fails on iOS 16 with this error:
/../../sdk/objc/unittests/RTCCameraVideoCapturerTests.mm:556: error: -[RTCCameraVideoCapturerTestsWithMockedCaptureSession testStartCaptureSetsOutputDimensionsInvalidPixelFormat] : ((width) equal to ([output.videoSettings[(id)kCVPixelBufferWidthKey] intValue])) failed: ("110") is not equal to ("0")
https://chromium-swarm.appspot.com/task?id=5fc0ac239b2dd110
Change-Id: Ia0a5c4290261b204d5e369dfc62113268ef48127
Bug: webrtc:14829, b/264630045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290895
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39130}
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}
