This ensures that the payload descriptor and potential generic
descriptors uses the same temporal layer.
Bug: b/200518293
Change-Id: I17e980b47fe6c814cb393fc459064576447aa27a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236520
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35275}
Holdback window can be specified as absolute time and in terms of packet
send times. Example:
WebRTC-TaskQueuePacer/Enabled,holdback_window:20ms,holdback_packet:3/
If current conditions have us running with 2000kbps pacing rate and
1250byte (10kbit) packets, each packet send time is 5ms.
The holdback window would then be min(20ms, 3*5ms) = 15ms.
The default is like before 1ms and packets no take into account when
TQ pacer is used, parameters have no effect with legacy process thread
pacer.
Bug: webrtc:10809
Change-Id: I800de05107e2d4df461eabaaf1ca04fb4c5de51e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233421
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35266}
This is somewhat klugey, because it does the same checks at two
different layers in the stack, in different functions, which runs
the risk of making them out of sync. But it solves the immediate
problem.
Bug: chromium:1249753
Change-Id: I2ad96f0cc9499c15540ff6946a409b40df3e3925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235826
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35259}
This CL includes two changes that break bit-exactness, but that haven't
changed the way AGC2 behaves - the new behavior has been verified with
audioproc_f on a collection of AEC dumps and Wav files (42 recordings
in total).
1) The fixed digital controller can directly be initialized in the
`GainController2` ctor. Before, `SetGainFactor()` was called after the
creation of the object and that caused an initial ramp up lasting one
10 ms frame from -inf to 0 dB. As an effect of the new initialization,
the initial ramp up doesn't happen anymore.
2) In [1] the AGC2 VAD has been moved from the adaptive digital
controller into `GainController2`. In order to not break bit-exactness,
the VAD was placed after the fixed digital controller and before the
adaptive digital one. However, to reduce the chance of incorrect
estimation of the speech probability, the VAD should analyze the
audio before any digital processing is applied inside AGC2.
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I9418229cbe537014fed8271c5550c3ce2bc88e26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235240
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35252}
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Id9849c4463791f5a203afe31efc163efb4d4458e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234583
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35248}
Only used in unit tests and a duplication of what `capture_output_rms_`
already does.
This CL also removes `AudioProcessingStats::output_rms_dbfs`, which is
now unused.
Bug: webrtc:5298
Fix: chromium:1261339
Change-Id: I6e583c11d4abb58444c440509a8495a7f5ebc589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235664
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35246}
To be able to generate candidates the configuration must support at
least one of the following:
* enable acknowledged bitrate as a candidate,
* enable delay based bwe as a candidate, or
* enabled a candidate factor other than 1.0.
If none of the above is supported then the configuration will be marked
as invalid and thus the `LossBasedBweV2` will be disabled.
Bug: webrtc:12707
Change-Id: I836ee59a396497f577b34fe0650ffc79f6cadc31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235210
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35239}
Also stop using ApplyConfig() and in [1] fix the build errors when
WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE is defined.
[1] modules/audio_processing/test/audio_processing_builder_for_testing.cc
Bug: webrtc:5298
Change-Id: I50dc5668b952e7ca7fa83c7a5182c013e928c450
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235365
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35228}
Not passing the sample rate to the `VoiceActivityDetectorWrapper` ctor
yet since that would require an unnecessary refactoring of `AdaptiveAgc`
which will soon be removed.
Instead, to ensure correct initialization until the child CL [1] lands,
`VoiceActivityDetectorWrapper::initialized_` is temporarily added.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234583
Bug: webrtc:7494
Change-Id: I4b4be7b8106ba36c958d91bf263a7b30271a1ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234587
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35213}
which can no longer happen since the end index and delta sizes are
checked in the surrounding condition.
Replace with a DCHECK to guard against potential errors.
BUG=None
Change-Id: I868d54c5923de773f248d10a40dbc6b2c563b0f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231957
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35210}
Internal refactoring of AGC2 to decouple the VAD, its wrapper and the
peak and RMS level measurements.
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Ib560f1fcaa601557f4f30e47025c69e91b1b62e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234524
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35208}
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized
and then the new config is applied. That is error prone and for example
breaks bit exactness in [1].
Changes:
- `GainController2` must be created by passing configuration,
sample rate and number of channels
- `GainController2::ApplyConfig()` removed
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234587.
Bug: webrtc:7494
Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35206}
This change improves echo canceller transparency by enabling the use
of a non-capped ERLE when computing the residual echo spectrum for
dominant nearend detection.
Experimentation has shown that the feature improves echo canceller
transparency and user ratings.
Implementation CL:
https://webrtc-review.googlesource.com/c/src/+/221920
Bug: webrtc:12870
Change-Id: I7dc66810e8300cd35321bcd5b9fae9bc3386836d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234841
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35186}
Sends a VideoLayersAllocation header extension if frame rate change more than 5fps since the last time it was sent with valid frame rate and resolution.
Bug: webrtc:12000
Change-Id: I2572c966025cc2c22743bbe2187cec7cceb86d01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234752
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35180}
Internal refactoring of AGC2. This CL is needed in preparation for its
child CL to correctly show the upcoming changes in the diff.
Bug: webrtc:7494
Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35170}
It seems the Android CTS tests only verify that 16x16 aligned resolutions
are supported.
This change checks the validity of input frame's size when initialing
or encoding processes are about to start using H/W MediaCodec.
This change has additional APIs to retrieve
|requested_resolution_alignment| and |apply_alignment_to_all_simulcast_layers|
from JAVA VideoEncoder class and its inherited classes. HardwareVideoEncoder
using MediaCodec has values of 16 and true for above variables.
Bug: webrtc:13089
Change-Id: I0c4ebf94eb36da29c2e384a3edf85b82e779b7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229460
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35169}
This avoids an additional step where we originally copied content from
PipeWire buffer to a temporary location and from there to DesktopFrame.
This results into less copy operations and hopefully to faster
screensharing.
I didn't do some exact measures, but simply running htop while sharing a
4k screen I can see following results (usage per top 5 processes):
1) Without this change - 66%, 64%, 26% 23%, 10%
2) With this change - 41%, 39%, 19%, 17%, 12%,
Bug: webrtc:13239
Change-Id: I6a661ecc96bfeef370c1a5a3b9dc5e3c0fc665c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231684
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35156}
This is a reland of f2177f6612079ccce9c320ea7e77bc934c684f5c
Original change's description:
> PipeWire capturer: implement proper DMA-BUFs support
>
> Currently both KWin (KDE) and Mutter (GNOME) window managers don't
> use DMA-BUFs by default, but only when client asks specifically for
> them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
> While current implementation works just fine on integrated graphics
> cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
> code either crashes or screensharing is slow and unusable.
>
> To fix this, DMA-BUFs has to be opened using OpenGL context and not
> being directly mmaped(). This implementation requires to use DMA-BUF
> modifiers, as they are now mandatory for DMA-BUFs usage.
>
> Documentation for this behavior can be found here:
> https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox
>
> Bug: chromium:1233417
> Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34889}
Bug: chromium:1233417
Change-Id: I308501d86ec18ab6df9bcee569c4b72df7926549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35152}
When new packets are enqueued after a dead-period where media debt is
zero, that time slice should not be used to reduce the debt for the
new packet.
Bug: webrtc:10809
Change-Id: Ifb960548e6aa349b79f37743cbfed78a5c937a13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234081
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35143}
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
Tested: compiled Chrome with this patch and made an appr.tc test call
Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
