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- /*
- * 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 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 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;
- // Robust VAD probability and speech decisions.
- constexpr float kDefaultSmoothedVadProbabilityAttack = 1.f;
- constexpr int kDefaultLevelEstimatorAdjacentSpeechFramesThreshold = 1;
- // Saturation Protector settings.
- constexpr float kDefaultInitialSaturationMarginDb = 20.f;
- constexpr float kDefaultExtraSaturationMarginDb = 2.f;
- 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_
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