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}
Original CL: https://webrtc-review.googlesource.com/c/src/+/101340
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It
is always less than or equal to 0. This CL changes inverts the value
logged. The noise level value should be somewhere between -90 and 0
dBFS.
The histogram description is updated in
https://chromium-review.googlesource.com/c/chromium/src/+/1264578
Bug: webrtc:7494
Change-Id: I0b53630d4284ce1078fd453e05e89ee53ca9f6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/104063
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25021}
Hypothetical scenario: short weak speech at start of call, then high
noise. The digital adaptive AGC2 would pick a high gain, and then
continue to apply it on the noise. Unless the noise is detected by the
noise estimator, the gain would never be reduced.
This CL addresses the issue by sending limiter gain info to the
adaptive digital AGC2.
Bug: webrtc:7494
Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64
Reviewed-on: https://webrtc-review.googlesource.com/102641
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24922}
This reverts commit 3a9731ff2f.
Reason for revert: Seems to cause crashes in Chrome browser tests, see for example https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8934487169011818016/+/steps/browser_tests__retry_with_patch_/0/logs/WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsOfferEcdsaAnswerEcdsa/0
Original change's description:
> Bug in histogram metric reporting.
>
> A (actually several weeks) while ago, we noticed an error with the
> WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
> the value 0. Here is why:
>
> The histogram bins go from 0 to 100. But the value logged is dBFS. It is
> always less than or equal to 0. This CL changes the bins.
>
> Bug: webrtc:7494
> Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
> Reviewed-on: https://webrtc-review.googlesource.com/101340
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24800}
TBR=saza@webrtc.org,aleloi@webrtc.org
Change-Id: I84883f73710b7e13aa90ee29b140acfc417f109f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7494
Reviewed-on: https://webrtc-review.googlesource.com/101701
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24809}
A (actually several weeks) while ago, we noticed an error with the
WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported
the value 0. Here is why:
The histogram bins go from 0 to 100. But the value logged is dBFS. It is
always less than or equal to 0. This CL changes the bins.
Bug: webrtc:7494
Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6
Reviewed-on: https://webrtc-review.googlesource.com/101340
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24800}
We add 2 metrics for measuring applied digital gain to
AgcManagerDirect. We also add an applied gain and an estimated noise
metric to Agc2.
Chromium histogram CL is
https://chromium-review.googlesource.com/c/chromium/src/+/1170833
Bug: webrtc:7494
Change-Id: Ie40873f9e43bc7d34d8f5473cd73bd47eb84e855
Reviewed-on: https://webrtc-review.googlesource.com/93468
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24290}
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}
If the adaptive gain is too low, we raise it slowly and only during
speech.
The CL gives better behavior at the start of a call. If the gain is too
high, the fixed-digital limits it. The gain is also quickly reduced by
the AdaptiveGainApplier.
Bug: webrtc:7494
Change-Id: I683f1e3e463cddec2d91f6c7f15c73e744430034
Reviewed-on: https://webrtc-review.googlesource.com/71484
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23053}
Added a new sub-module 'GainApplier'. The build target is
'modules/audio_processing/agc2:gain_applier'. A small refactoring
makes the GainApplier used in adaptive-digital AGC2.
The AGC2 now multiplies samples with a gain in 3 places. It's the
GainApplier, the GainCurveApplier, and the FixedGainController. The
GainApplier is used in AdaptiveDigitalGainApplier and will be used as
a pre-amplifier.
Bug: webrtc:9138
Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6
Reviewed-on: https://webrtc-review.googlesource.com/69321
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22849}
AGC2 component that computes and applies the digital gain.
The gain is computed from an estimated speech and noise level.
This component decides how fast the gain can change and what it
should be.
Bug: webrtc:7494
Change-Id: If55b6e5c765f958e433730cd9e3b2b93c14a7910
Reviewed-on: https://webrtc-review.googlesource.com/64985
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22741}
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}
