The extra saturation margin is a setting for the SaturationProtector
in GainController2. The higher it is, the less gain GC2 will apply. In
this CL we pipe the setting up to audio_processing.h. Now the setting
can be set at a high level.
Also in this CL add a few (missing, they should have been there
already) tests for the GC2 and GC2 with saturation margin.
Bug: webrtc:7494
Change-Id: I1b61f1662e6c6a8817fd5b0e845339694bf8d50d
Reviewed-on: https://webrtc-review.googlesource.com/c/109001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25470}
The AdaptiveAgc often boosts the signal outside of Float S16 range. It
is expected, which is why we have a limiter after it in the process
chain. But it turns out that this happens regularly even for simple
input examples. The output signal peaks can be as high as +4 dBFs for a
single speaker example (which should be easy). It leads to excessive
gain modulation by the limiter.
This CL is a new tuning designed to produce a safer gain. After this,
we shouldn't hit the saturation region of the limiter as often. But we
will still maintain a high gain.
We have a 'configurable kill-switch': the settings can be changed via
field trials WebRTC-Audio-Agc2Force(Initial|Extra)SaturationMargin.
Bug: webrtc:7494, chromium:892043
Change-Id: I5014377050c74c32ae8998282991141eae31cf58
Reviewed-on: https://webrtc-review.googlesource.com/c/102922
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25006}
1. Adds support for Reset calls in AGC2. The AGC will be reset during
analog gain changes.
2. Allows AdaptiveModeLevelEstimator to return estimates > 0. This can
happen if the signal gain is too high. It's needed for letting the
analog AGC know that the gain is too high.
Bug: webrtc:7494
Change-Id: I38def17c21cc01c36aaea79a2401d8c2f289407b
Reviewed-on: https://webrtc-review.googlesource.com/79360
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23805}
* Move 'VadWithLevel' to AGC2 where it belongs.
* Remove the vectors from VadWithLevel. They were there to make it work
with modules/audio_processing/vad, which we don't need any longer.
* Remove the vector handling from AGC2. It was spread out across
AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests.
* Hack the RNN VAD into VadWithLevel. The main issue is the resampling.
Bug: webrtc:9076
Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e
Reviewed-on: https://webrtc-review.googlesource.com/77364
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23688}
Another submodule of the Automatic Gain Controller 2. It refines the
biased estimate of the Adaptive Mode Level Estimator. It works by
generating a delayed stream of peak levels. The delayed peaks are
compared to the level estimate.
Bug: webrtc:7494
Change-Id: If4c2c19088d1ca73fb93511dad4e1c8ccabcaf03
Reviewed-on: https://webrtc-review.googlesource.com/65461
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22732}
The level estimator (AdaptiveModeLevelEstimator) produces a biased
estimate of the speech level. In our model, we use another module
(the SaturationProtector) to compute the bias. This CL contains the
estimator and a stub of the saturation protector.
Bug: webrtc:7494
Change-Id: I0df736d0346063f544fa680b4cc84177ea548545
Reviewed-on: https://webrtc-review.googlesource.com/64820
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22641}
