The function has been deprecated in favor of
FrameEncodeMetadataWriter::FillMetadataAndTimingInfo().
Bug: chromium:328598314
Change-Id: Iaf2008e855dbd71f2d7cf412d95c5932b3645d71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356042
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42613}
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}
Similar to BGRA/RGBA we added recently, formats from PipeWire are in
big-endian, while WebRTC (using libyuv) is little-endian, therefore we
have to map BGR to RGB and not RGB to RGB as colors would be off. Also
add some additional formats supported by libyuv.
Bug: webrtc:42225999
Change-Id: Iee8303f0922fe434069b2b3f88994abecf7d2cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355860
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42609}
* 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}
Only minor positive changes seen in initial testing. Let's default-
enable and monitor behavior through the normal release cycle.
Bug: b/349561566
Change-Id: Id6b39daa159068bf076acc34888b5d7eaf110329
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356641
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42607}
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).
Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).
Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
Before this change the AV1 encoder wrapper converted target frame rate from double to integer with rounding to the middle. That approach resulted in a bitrate mismatch caused by rounding error. The mismatch was especially high at low frame rates. For example, at target frame rate 1.4fps the bitrate mismatch reached 40%:
out/debug/video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --framerate_fps=1.4 --width=320 --height=180 --bitrate_kbps=32 --num_frames=600
...
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {39.171875,0} n%
After the change the mismatch reduced to ~2% in the same scenario:
RESULT s0t0_bitrate_mismatch_pct: DISABLED_EncodeDecode= {-2.178125,0} n%
Bug: b/337757868
Change-Id: Ia51f92b3dfdce103eed1d04cac0e084b69fa8213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42601}
There's some test code associated with this code path that can
be deleted, so this is a first step towards removing it. From what
I can tell, this is never used.
Bug: none
Change-Id: Idfb8a6c58b929c2eedd0cfc7bdc72f5b3862f5bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356481
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42600}
This is cherry-picked from WebKit's patch for fixing a fuzzer failure.
The original patch: https://github.com/WebKit/WebKit/pull/30438
Bug: chromium:41480904
Change-Id: Ic8eddb9de816c4c8d720dac6d4c55d1db3f0596e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#42598}
in addition to last received frame rtp timestamp. This helps in cases
where frames continue to be received but the decoder fails to decode.
BUG=None
Change-Id: I56ad5f9ef85cc598d3c1a1971c4c697eb6b70165
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356080
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42596}
This can be tested by trying to connect to a TURN server that does not
listen on a specific TCP port.
BUG=None
Change-Id: I7029112afa4b1b4376220dfc2d613a30090e4f7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42595}
If traffic policing is enforced by dropping packets, RTT can still be low.
If a packet is dropped that is needed to contninue decoding, it make sense that a nack request is sent until the packet is received, or a new key frame is requested. A key frame will be requested after 3s.
For now, this cl only increase the number of times a packet can be requested.
Bug: b/317178411
Change-Id: Iea75d36ed06f346af1dd4e55a9961d5eca45f519
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356482
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42594}
Behind a flag, the new behavior changes how the "media rate" utilization
is calculated:
* Instead of per spatial & temporal layer, it's per spatial layer only.
* Overshoot is compared to real target vs adjusted target.
* Window takes quite periods/frame drops more into consideration.
This should lead to less push-back when not network constrained and
complex content is used causing bursty behavior.
Bug: b/349561566
Change-Id: I402e6531183493c963fec48ae363ce0b859b396a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356480
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42593}
This class measures the allocated cumulative byte budget (as specified
by one or more rate updates) and the actual cumulative number of bytes
produced over a sliding window.
A utilization factor (produced bytes / budgeted bytes) is calculated
seen from the first data point timestamp until the last data point
timestamp plus the amount time needed to send that last data point
given no further updates to the rate.
Wireup to EncoderBitrateAdjuster will happen in a follow-up CL.
Bug: b/349561566
Change-Id: Id0dc183b07a96366531007be9ff1c1ec6574e9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356200
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42591}
This helps with making AudioBuffer compatible with current and upcoming
code that uses audio_views.h (a simpler abstraction).
Bug: chromium:335805780
Change-Id: Ib59bba274c7abfb441e3c4d606f804b365df236d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355844
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42590}
Bitrate limits should have been applied to the spatial layers in ApplySpatialLayerBitrateLimits (and usage is restricted to a single active stream/layer).
Bug: b/299588022
Change-Id: Iaae4ece28b8a95eea7d4bacba292847ba5b4000b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355841
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42588}
This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.
Instead of InitializeIfNeeded:
* Offer a way to explicitly initialize PushResampler via the ctor
(needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
(All calls to Resample() were preceded by a call to Initialize)
As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.
Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.
Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
Trying to fix the error:
bad request: build: constraints for webrtc:try not found
Change-Id: Icf96d5082ce09a60d079b91117a3786ddb97269c
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356301
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42585}
Tried 'led' today and it resulted in the below error:
rpc error: code = PermissionDenied desc = user does not have permission "buildbucket.builds.create"
Change-Id: I361859b6f6ee58a67ac08e615cb88761fb39d67e
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356300
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42583}
Some dependencies still exist but are a bit more complex to remove.
This CL removes either unused or easily replaced with ToString()
instances of ostream usage. In one case, moving the operator<<
implementation to the one test file that requires it.
Bug: webrtc:8982
Change-Id: Ia5c840b12a42893494af401317a3daf2fe50ba9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356240
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42582}
Also adding more checks around parameter valuesm since the
`sample_rate_hz` argument is technically not needed, but will be
removed in a follow-up CL.
Bug: chromium:335805780
Change-Id: Ia7e50658f8a686ab71980f9c59cce5f097b0af40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353340
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42581}
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}
* FrameCombiner is simpler. No additional channel pointers for buffers.
* Improve consistency in using views in downstream classes.
* Deprecate older methods (some have upstream dependencies).
* Use samples per channel instead of sample rate where the former is
really what's needed.
Bug: chromium:335805780
Change-Id: I0dde8ed7a5a187bbddd18d3b6c649aa0865e6d4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352582
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42575}
The temporal id must be read from `EncodedImage` rather than codec
specifics for AV1. Furthermore, in some configs the spatial id of
`EncodedImage` is populated and set to 0 while the simulcast id can
also be simultaneously populated and set to values, including non-zero.
To solve this, just take the max of the two.
Bug: b/349561566
Change-Id: I46c61b7f0fff7a7ab8d7262c3a8d413f49b3286a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355904
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42573}
The new constructor exposes an already existing constructor,
and is used to create a (test) UDPPort
with a socket...so that one does not (really) need a
socket factory.
Bug: b/339018639
Change-Id: Ib591fe6ae61519fe29cdea819192694448b071e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42571}
Add default values and a static Create() function that determines
the parameters to use based on the specified codec and potential
field trial overrides.
Bug: chromium:328598314
Change-Id: I7a9331a1fd0ed4bd258788760592ea84e535e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355903
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42567}
