This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
In preparation for upcoming changes in GetSimulcastConfig(), which will require a vector of stream resolutions instead of just the max resolution as an input, switch tests to use CreateEncoderStreams() instead of calling GetSimulcastConfig() directly.
Bug: webrtc:351644568, b/352504711
Change-Id: I541dd54a21a8b75028cff07a250f858a47898223
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357400
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42648}
Merge GetAndSetRtpSendParametersScaleResolutionDownByVP8, GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution, GetAndSetRtpSendParametersScaleResolutionDownByH264 and GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution into one parameterized test.
PS. Not sure why we need separate tests for VP8 and H264. Underlaying code paths are codec agnostic as I can see.
Bug: webrtc:351644568, b/352504711
Change-Id: Ic95c59c1bfacdba8a42de8ecca9cab42014842e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42646}
This is a cleanup of simulcast.cc. max_qp is not needed to decide simulcast config. Move setting of max QP in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams(), where it can be set per stream.
Bug: webrtc:351644568, b/352504711
Change-Id: Ia0e3e9d90032383574dc8867b30d362e9c5df7e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42634}
This is a cleanup of simulcast.cc. bitrate_priority is not needed to decide simulcast config. Move setting of bitrate priority in VideoStream one level up, to EncoderStreamFactory::CreateEncoderStreams().
Bug: webrtc:351644568
Change-Id: I002d728ccf8d141fe4bbb32b390129ce57c830cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357101
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42629}
This CL adds the sequence checks to the functions that creates or
destroys the encoder context. Those functions must be executed on
the webrtc encoder sequence.
Bug: b/320555128
Change-Id: I1daa93f2f5326073e8d75e1d711d7554bed76a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Hirokazu Honda <hiroh@google.com>
Cr-Commit-Position: refs/heads/main@{#42612}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
Currently, if the single-encoder mode fails to initialize, the
callback is cleared on the encoder with Release() call.
Below, the encoder_context for the first stream will be reused but
then we only intercept the callback for the stream_idx>0.
Therefore if RegisterEncodeCompleteCallback() is called before the InitEncode(), the first stream will end up with nullptr callback.
To ensure this doesn't happen, restore the callback on the reusable encoder.
Bug: none
Change-Id: I1c830f3bf71f64807d5cc1a3000b73834011bde4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42580}
Libvpx works without this, so the existing tests pass. However, other
encoder implementations (like rtc_video_encoder in Chrome) look at
different fields and get confused about the configuration.
Test: Integration tests with Chrome and windows hardware encoders.
Bug: webrtc:348342168
Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42555}
and move usages to webrtc::RefCountInterface
This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.
Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
The encoders stored in SimulcastEncoderAdapter might have a reference
which is passed by the encoder factory. Therefore, the encoder factory
needs to outlive the encoders.
A class inheriting SimulcastEncoderAdapter possibly owns the resource.
Since a base class' destructor cannot be called before a child class,
the child class needs to destroy encoders by invoking
DestroyStoredEncoders() in the destructor.
Bug: b:320555128
Test: Build
Test: unit test
Change-Id: I34672556bab5f6b0f754edb840e9eb42d38f292e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351442
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Hirokazu Honda <hiroh@google.com>
Cr-Commit-Position: refs/heads/main@{#42436}
This is in preparation for moving the assignment of PT numbers
to a different place.
Bug: webrtc:42226302
Change-Id: I821f2b6b15eb9f9f3715714e2c2c220c5e5a219e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352921
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42422}
reduced-size RTCP, i.e. not prefixing RTCP packets with either a sender report or receiver report has been implemented for a long time but only for video.
This CL adds it for audio as well. This reduces the size of audio NACKs (16 bytes, typically one NACK per packet) sent by not prefixing it with a receiver report (32 bytes).
Other packets are not affected as e.g. transport-cc feedback does not add a RR even though that is technically required.
The effect on NACK can be tested by running Chromium with
--disable-webrtc-encryption --force-fieldtrials=WebRTC-FakeNetworkReceiveConfig/loss_percent:5/
against this fiddle negotiating audio nack:
https://jsfiddle.net/fippo/8ubtLnfx/1/
BUG=webrtc:340041654
Change-Id: I06fb94742ff1b6f9a464c404bfc53913f23498d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350269
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42330}
So that this class can use propagated field trials instead of the global
Bug: webrtc:42220378
Change-Id: Ic1dba0c4967735606904329f7e9e6c09f186b809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42326}
This is done in case the new source has a different clock source which
might be out of sync with the previous one, leading to timestamp
conflicts.
Bug: b/340198307
Change-Id: Id1a82823ca85e03d99dad31b7ce9f19f09447755
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42312}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
This relands the
https://webrtc-review.googlesource.com/c/src/+/349920
where patchset#1 is identical to the original change,
patchset#2 undoes (postpones) the expectation downstream propagates the Environment too.
Bug: webrtc:42220378
Change-Id: I4a9a32bb0926a875d37f3ba19dd5309e97546553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350364
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42298}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
Bug: webrtc:42220378
Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42288}
Replace factory that takes optional FieldTrialView with a constructor that takes non-optional reference to the same interface - all callers already guarantee it is not nullptr
Replace several local IsEnabled/IsDisabled helpers with the same helpers in FieldTrialView
In CongestionWindowPushbackController tests pass field trials bypassing global field trial string
Bug: webrtc:42220378
Change-Id: Ic49ad78919d834a5e3b9b69545d3b39088023a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349900
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42270}
Instead of from the global field trial string.
Bug: webrtc:42220378
Change-Id: Iddb41429e388792de02f702b4caa35689c57d9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347720
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42201}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
Instead of passing it as optional parameter during construction, pass field trials as required parameters on use.
Test that create the EncoderStreamFactory might not have an easy access to the actual field trials, but prod code has appropriate field trials when uses the factory.
This way EncoderStreamFactory doesn't need to depend on global field trial string through FieldTrialBaseConfig class.
Bug: webrtc:10335
Change-Id: I8f7030e41579ff2c5dd362c491a4e1624b23e690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42098}
following the audio changes. Note that RTT-related fields require
DLRR and are not implemented yet.
BUG=webrtc:12529
Change-Id: I3f9449fbe876a1b282a32f2bcebe1cf3e10989bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346580
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42069}
To pass field trials to EncoderStreamFactory in FakeVideoSendStream and thus reduce dependency on the global field trial.
Bug: webrtc:10335
Change-Id: Iad32881c2d9158fe1d77f1b71f8d606374ea111e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42023}
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.
The old fields are preserved for compatibility with downstream projects, but will be removed in the future.
Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
When an error occurs, the callback needs to be invoked or the
signaling thread may block indefinitely waiting for it.
Bug: webrtc:15871
Change-Id: Ib73382aff07b3632794300985223c70c24f554f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342901
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41904}
A call to GetScalabilityMode was added for logging purpose and causes an expectation failure for tests using 4 temporal layers.
Plan is to remove the old GetScalabilityMode and keep only the one that returns an optional.
Change-Id: I0e37a496bb621d9754d6572ef5838b58193aa183
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341520
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41838}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
which needs to be added to the remote codecs a=fmtp:
This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.
This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.
BUG=webrtc:10107
Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.
Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41787}
