8aaa604375
Bug: webrtc:7494 Change-Id: Id288dd426e1c2754805bc548fbffe0eaeaacf3da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213420 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33605}
89 lines
3.6 KiB
C++
89 lines
3.6 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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#include <stddef.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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constexpr int kFrameDurationMs = 10;
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constexpr int kSubFramesInFrame = 20;
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constexpr int kMaximalNumberOfSamplesPerChannel = 480;
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constexpr float kAttackFilterConstant = 0.0f;
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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 = 0.9f;
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// The amount of 'memory' of the Level Estimator. Decides leak factors.
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constexpr int kFullBufferSizeMs = 1200;
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constexpr float kFullBufferLeakFactor = 1.0f - 1.0f / kFullBufferSizeMs;
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constexpr float kInitialSpeechLevelEstimateDbfs = -30.0f;
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// Robust VAD probability and speech decisions.
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constexpr int kDefaultVadRnnResetPeriodMs = 1500;
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static_assert(kDefaultVadRnnResetPeriodMs % kFrameDurationMs == 0, "");
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constexpr float kDefaultSmoothedVadProbabilityAttack = 1.0f;
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constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 1;
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// Saturation Protector settings.
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constexpr float kDefaultInitialSaturationMarginDb = 20.0f;
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constexpr float kDefaultExtraSaturationMarginDb = 2.0f;
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constexpr int kPeakEnveloperSuperFrameLengthMs = 400;
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static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
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"Full buffer size should be a multiple of super frame length for "
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"optimal Saturation Protector performance.");
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constexpr int kPeakEnveloperBufferSize =
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kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
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// This value is 10 ** (-1/20 * frame_size_ms / satproc_attack_ms),
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// where satproc_attack_ms is 5000.
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constexpr float kSaturationProtectorAttackConstant = 0.9988493699365052f;
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// This value is 10 ** (-1/20 * frame_size_ms / satproc_decay_ms),
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// where satproc_decay_ms is 1000.
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constexpr float kSaturationProtectorDecayConstant = 0.9997697679981565f;
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// This is computed from kDecayMs by
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// 10 ** (-1/20 * subframe_duration / kDecayMs).
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// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
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// kDecayMs is defined in agc2_testing_common.h
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constexpr float kDecayFilterConstant = 0.9998848773724686f;
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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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