Made path from NetEq to AudioTransport ready for many-channel audio.
If there is one stream, we can handle anything that fits in an
AudioFrame. For many streams, the current limit is 6.
Some multi-channel combinations are not supported: e.g. if we get
stereo audio and attempt to play out 6 channels.
Changes:
* AudioFrameOperations - replaced the MonoTo* and *ToMono methods by
UpmixChannels & DownmixChannels.
* AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing
algorithm to handle many channels.
Bug: webrtc:8649
Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106040
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26446}
This CL refactors AGC2 and fixes the order with which the fixed
and the adaptive digital gain controllers are applied - i.e., fixed
first, then adaptive and finally limiter.
FixedGainController has been removed since we need to split the
processing done by the gain applier and the limiter.
Also, GainApplier and Limiter are easy enough to be used without
a wrapper and a wrapper would need 2 separated calls in the right
order - i.e., error prone.
FrameCombiner in audio mixer has been adapted and now only uses the
limiter (which is what is needed since no gain is applied).
The unit tests for FixedGainController have been moved to
gain_controller2_unittests. They have been re-adapted and
ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned.
Bug: webrtc:7494
Change-Id: I4d7daeae917257ac019a645b74deba6642f77322
Reviewed-on: https://webrtc-review.googlesource.com/c/108624
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25477}
The FixedGainController is used in two places.
One is the AudioMixer. There it's used to limit the audio level after
adding streams. The other is GainController2, where it's placed after
steps that could boost the audio level outside the allowed range.
We log metrics from the FGC. To avoid confusion, this CL makes the two
use cases log to different histograms.
Chromium histogram CL is
https://chromium-review.googlesource.com/c/chromium/src/+/1170833
Bug: webrtc:7494
Change-Id: I1abe60fd8e96556f144d2ee576254b15beca1174
Reviewed-on: https://webrtc-review.googlesource.com/93464
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24284}
The FrameCombiner sub-module of the AudioMixer uses one of two
limiters. One is an AudioProcessingModule with AGC1 enabled and
configured as a limiter. The other is the limiter part of AGC2. This
change removes the APM-AGC1 limiter. This requires small changes to
FrameCombiner, AudioMixerImpl and tests.
We also stop using the finch experiment flag.
Bug: webrtc:8925
Change-Id: Id7b8349ec4720b6417b15eaf70ed1a850b6ddbed
Reviewed-on: https://webrtc-review.googlesource.com/84620
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23727}
We update the configuration settings for AGC2. We also update their
effects. Now, 'gain_controller2.enable=true' means 'first run Adaptive
AGC2; then run AGC2 limiter'.
Previously, only the AGC2 limiter was implemented. To run that, one
had to set both 'gain_controller2.enable=true' and
'gain_controller2.enable_limiter=true'.
This setting also enables adaptive AGC2 in the test tool 'audioproc_f'.
Bug: webrtc:7494
Change-Id: I0d5dfe443f2cdc0ecf3aa4054442dab6276d284d
Reviewed-on: https://webrtc-review.googlesource.com/64990
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22669}
And fix typo in UMA metric.
We have this pattern in the FrameCombiner component of the AudioMixer:
if (number_of_streams <= 1) {
// Copy or fill with zeros.
return;
}
// Mix and limit
LogMixingStats(/* args */);
When there is only one remote stream, info about active streams and
sample rate is not logged. This CL moves the call to log stats before
the 'return'.
Bug: webrtc:8925
Change-Id: I7b54f61f628273631909dafbfafa21e155e18d4a
Reviewed-on: https://webrtc-review.googlesource.com/62860
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22493}
We want to know how the AudioMixer is used and how FixedGainController
behaves.
The WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.* metrics measures
how often the input level hits different regions of the Fixed Gain
Controller gain curve (when the limiter is enabled). They also measure
how long the metrics stay in different regions. They are related to
WebRTC.Audio.ApmCaptureOutputLevelPeakRms, but the new metrics measure
the level before any processing done in APM.
The AudioMixer mixes incoming audio streams. Their number should be
mostly constant, and often some of them could be muted. The metrics
WebRTC.Audio.AudioMixer.NumIncomingStreams,
WebRTC.Audio.AudioMixer.NumIncomingActiveStreams log the number of
incoming stream and how many are not muted. We currently don't have
any stats related to that.
The metric WebRTC.Audio.AudioMixer.MixingRate logs the rate selected
for mixing. The rate can sometimes be inferred from
WebRTC.Audio.Encoder.CodecType. But that metric measures encoding and
not decoding, and codecs don't always map to rates.
See also accompanying Chromium CL
https://chromium-review.googlesource.com/c/chromium/src/+/939473
Bug: webrtc:8925
Change-Id: Ib1405877fc1b39e5d2f0ceccba04434813f20b0d
Reviewed-on: https://webrtc-review.googlesource.com/57740
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22443}
Allows mixing up to 4 input streams. Useful for profiling and manual
tests. Allows testing different combinations of input/output rates and
number of channels. Reads and writes WAV files. Can also configure
whether to use the Limiter component of the AudioMixer.
Bug: webrtc:8925
Change-Id: Iaf4fee5284980f6ed01f4bb721e49bb1af8dd392
Reviewed-on: https://webrtc-review.googlesource.com/56842
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22209}
The webrtc::AudioMixer uses a limiter component. This CL allows
changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome
field trial.
The AGC2 limiter has a float interface. We plan to eventually switch
to the AGC2 limiter. Therefore, we will now mix in de-interleaved
floats. Float mixing will happen both when using the old limiter and
when using the new one.
After this CL the mixer will support two limiters. The limiters have
different interfaces and need different processing steps. Because of
that, we make (rather big) changes to the control flow in
FrameCombiner. For a short while, we will mix in deinterleaved floats
when using any limiter.
Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/
Reverted in https://webrtc-review.googlesource.com/c/src/+/57940
because of both breaking compilation and having a severe error. The
error is fixed and a test is added. The compilation issue is fixed.
Bug: webrtc:8925
Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0
Reviewed-on: https://webrtc-review.googlesource.com/58085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22207}
This reverts commit bd7b461f16b53363e2c510893d8d3aade5737f3a.
Reason for revert: Broke the internal project. The issue maybe related to the apm_debug_dump configuration.
Original change's description:
> Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller.
>
> The webrtc::AudioMixer uses a limiter component. This CL changes the
> APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
>
> The new limiter has a float interface. Since we're moving to it, we
> now mix in floats as well. After this CL the mixer will support two
> limiters. The limiters have different interfaces and need different
> processing steps. Because of that, we make (rather big) changes to the
> control flow in FrameCombiner. For a short while, we will mix in
> deinterleaved floats when using any limiter.
>
> NOTRY=true
>
> Bug: webrtc:8925
> Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
> Reviewed-on: https://webrtc-review.googlesource.com/56141
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22185}
TBR=gustaf@webrtc.org,aleloi@webrtc.org
Change-Id: I3dd1a2b1fca32c4dd046e6fc325744079e3ac5ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8925
Reviewed-on: https://webrtc-review.googlesource.com/57940
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22189}
The webrtc::AudioMixer uses a limiter component. This CL changes the
APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
The new limiter has a float interface. Since we're moving to it, we
now mix in floats as well. After this CL the mixer will support two
limiters. The limiters have different interfaces and need different
processing steps. Because of that, we make (rather big) changes to the
control flow in FrameCombiner. For a short while, we will mix in
deinterleaved floats when using any limiter.
NOTRY=true
Bug: webrtc:8925
Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
Reviewed-on: https://webrtc-review.googlesource.com/56141
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22185}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
