89 lines
3.5 KiB
C++
89 lines
3.5 KiB
C++
/*
|
|
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
|
|
|
|
#include <stddef.h>
|
|
|
|
namespace webrtc {
|
|
|
|
constexpr float kMinFloatS16Value = -32768.f;
|
|
constexpr float kMaxFloatS16Value = 32767.f;
|
|
constexpr float kMaxAbsFloatS16Value = 32768.0f;
|
|
|
|
constexpr size_t kFrameDurationMs = 10;
|
|
constexpr size_t kSubFramesInFrame = 20;
|
|
constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
|
|
|
|
constexpr float kAttackFilterConstant = 0.f;
|
|
|
|
// Adaptive digital gain applier settings below.
|
|
constexpr float kMaxGainChangePerSecondDb = 3.f;
|
|
constexpr float kMaxGainChangePerFrameDb =
|
|
kMaxGainChangePerSecondDb * kFrameDurationMs / 1000.f;
|
|
constexpr float kHeadroomDbfs = 1.f;
|
|
constexpr float kMaxGainDb = 30.f;
|
|
constexpr float kInitialAdaptiveDigitalGainDb = 8.f;
|
|
// At what limiter levels should we start decreasing the adaptive digital gain.
|
|
constexpr float kLimiterThresholdForAgcGainDbfs = -kHeadroomDbfs;
|
|
|
|
// This parameter must be tuned together with the noise estimator.
|
|
constexpr float kMaxNoiseLevelDbfs = -50.f;
|
|
|
|
// This is the threshold for speech. Speech frames are used for updating the
|
|
// speech level, measuring the amount of speech, and decide when to allow target
|
|
// gain reduction.
|
|
constexpr float kVadConfidenceThreshold = 0.9f;
|
|
|
|
// The amount of 'memory' of the Level Estimator. Decides leak factors.
|
|
constexpr size_t kFullBufferSizeMs = 1200;
|
|
constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
|
|
|
|
constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
|
|
|
|
// Saturation Protector settings.
|
|
float GetInitialSaturationMarginDb();
|
|
float GetExtraSaturationMarginOffsetDb();
|
|
|
|
constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
|
|
static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
|
|
"Full buffer size should be a multiple of super frame length for "
|
|
"optimal Saturation Protector performance.");
|
|
|
|
constexpr size_t kPeakEnveloperBufferSize =
|
|
kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
|
|
|
|
// This value is 10 ** (-1/20 * frame_size_ms / satproc_attack_ms),
|
|
// where satproc_attack_ms is 5000.
|
|
constexpr float kSaturationProtectorAttackConstant = 0.9988493699365052f;
|
|
|
|
// This value is 10 ** (-1/20 * frame_size_ms / satproc_decay_ms),
|
|
// where satproc_decay_ms is 1000.
|
|
constexpr float kSaturationProtectorDecayConstant = 0.9997697679981565f;
|
|
|
|
// This is computed from kDecayMs by
|
|
// 10 ** (-1/20 * subframe_duration / kDecayMs).
|
|
// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
|
|
// kDecayMs is defined in agc2_testing_common.h
|
|
constexpr float kDecayFilterConstant = 0.9998848773724686f;
|
|
|
|
// Number of interpolation points for each region of the limiter.
|
|
// These values have been tuned to limit the interpolated gain curve error given
|
|
// the limiter parameters and allowing a maximum error of +/- 32768^-1.
|
|
constexpr size_t kInterpolatedGainCurveKneePoints = 22;
|
|
constexpr size_t kInterpolatedGainCurveBeyondKneePoints = 10;
|
|
constexpr size_t kInterpolatedGainCurveTotalPoints =
|
|
kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
|