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d66a60597d
Add the option to run the adaptive digital controller of AGC2 without side-effects - i.e., no gain applied. Tested: adapation verified during a video call in chromium Bug: webrtc:7494 Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33875}
71 lines
2.9 KiB
C++
71 lines
2.9 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
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namespace webrtc {
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constexpr float kMinFloatS16Value = -32768.0f;
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constexpr float kMaxFloatS16Value = 32767.0f;
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constexpr float kMaxAbsFloatS16Value = 32768.0f;
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// Minimum audio level in dBFS scale for S16 samples.
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constexpr float kMinLevelDbfs = -90.31f;
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constexpr int kFrameDurationMs = 10;
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constexpr int kSubFramesInFrame = 20;
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constexpr int kMaximalNumberOfSamplesPerChannel = 480;
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// Adaptive digital gain applier settings below.
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constexpr float kHeadroomDbfs = 1.0f;
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constexpr float kMaxGainDb = 30.0f;
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constexpr float kInitialAdaptiveDigitalGainDb = 8.0f;
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// At what limiter levels should we start decreasing the adaptive digital gain.
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constexpr float kLimiterThresholdForAgcGainDbfs = -kHeadroomDbfs;
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// This is the threshold for speech. Speech frames are used for updating the
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// speech level, measuring the amount of speech, and decide when to allow target
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// gain reduction.
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constexpr float kVadConfidenceThreshold = 0.95f;
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// Adaptive digital level estimator parameters.
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// Number of milliseconds of speech frames to observe to make the estimator
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// confident.
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constexpr float kLevelEstimatorTimeToConfidenceMs = 400;
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constexpr float kLevelEstimatorLeakFactor =
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1.0f - 1.0f / kLevelEstimatorTimeToConfidenceMs;
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// Robust VAD probability and speech decisions.
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constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 12;
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// Saturation Protector settings.
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constexpr float kSaturationProtectorInitialHeadroomDb = 20.0f;
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constexpr float kSaturationProtectorExtraHeadroomDb = 5.0f;
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constexpr int kSaturationProtectorBufferSize = 4;
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// Set the initial speech level estimate so that `kInitialAdaptiveDigitalGainDb`
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// is applied at the beginning of the call.
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constexpr float kInitialSpeechLevelEstimateDbfs =
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-kSaturationProtectorExtraHeadroomDb -
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kSaturationProtectorInitialHeadroomDb - kInitialAdaptiveDigitalGainDb -
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kHeadroomDbfs;
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// Number of interpolation points for each region of the limiter.
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// These values have been tuned to limit the interpolated gain curve error given
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// the limiter parameters and allowing a maximum error of +/- 32768^-1.
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constexpr int kInterpolatedGainCurveKneePoints = 22;
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constexpr int kInterpolatedGainCurveBeyondKneePoints = 10;
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constexpr int kInterpolatedGainCurveTotalPoints =
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kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
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