This CL also includes the following changes:
- `AudioProcessing::Config::GainController2::noise_estimator`
deprecated
- `EnergyToDbfs()` optimized by removing unnecessary `sqrt`
- Unit test minor fix, incorrect type was used
Bug: webrtc:7494
Change-Id: I88a6672d6f7cd03fcf6a3031883522d256880140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230940
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34893}
- Bug fix: the desired initial gain quickly dropped to 0 dB hence
starting a call with a too low level
- New tuning to make AGC2 more robust to VAD mistakes
- Smarter max gain increase speed: to deal with an increased threshold
of adjacent speech frames, the gain applier temporarily allows a
faster gain increase to deal with a longer time spent waiting for
enough speech frames in a row to be observed
- Saturation protector isolated from `AdaptiveModeLevelEstimator` to
simplify the unit tests for the latter (non bit-exact change)
- AGC2 adaptive digital config: unnecessary params deprecated
- Code readability improvements
- Data dumps clean-up and better naming
Bug: webrtc:7494
Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33736}
The current noise level estimator has a bug due to which the estimated
level decays to the lower bound in a few seconds when speech is observed.
Instead of fixing the current implementation, which is based on a
stationarity classifier, an alternative, lightweight, noise floor
estimator has been added and tuned for AGC2.
Tested on several AEC dumps including HW mute, music and fast talking.
Bug: webrtc:7494
Change-Id: Iae4cff9fc955a716878f830957e893cd5bc59446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214133
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33733}
Done in preparation for the child CL which adds an alternative
implementation.
Bug: webrtc:7494
Change-Id: I4963376afc917eae434a0d0ccee18f21880eefe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214125
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33646}
Code style improvements done in preparation for a bug fix (TODO added)
which requires changes in the unit tests.
Note that one expected value in the unit tests has been adjusted since
the white noise generator is now instanced in each separate test and
therefore, even if the seed remained the same, the generated sequences
differ.
Bug: webrtc:7494
Change-Id: I497513b84f50b5c66cf6241a09946ce853eb1cd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214122
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33636}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
We put back the old noise estimator from LevelController. We add a few
new unit tests. We also re-arrange the code so that it fits with how
it is used in AGC2. The differences are:
1. The NoiseLevelEstimator is now fully self-contained.
2. The NoiseLevelEstimator is responsible for calling SignalClassifier
and computing the signal energy. Previously the signal type and
energy were used in several places. It made sense to compute the
values independently of the noise calculation.
3. Re-initialization doesn't have to be done by the caller.
4. The interface is AudioFrameView instead of AudioBuffer.
# Bots are green, nothing should break internal stuff
NOTRY=True
Bug: webrtc:7494
Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d
Reviewed-on: https://webrtc-review.googlesource.com/66380
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22738}
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.
Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
1. Level Estimator - it gets the energy and a speech probability
and updates a speech level estimate.
2. Noise Estimator - it gets an immutable view of the speech frame
and updates the noise level estimate
3. Gain applier - it gets the speech frame, the current speech and
noise estimates, and the speech probability. It finds a gain to
apply and applies it.
4. AdaptiveAgc - sets up and controls the sub-modules described
above.
Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
