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2ffafa8244
This CL does the following:
1. Adds a new AdaptiveModeLevelEstimatorAgc implementation of the Agc
interface. The new implementation differs from webrtc::Agc by
1. using the AGC2 speech level estimator in
GetRmsErrorDb. webrtc::Agc implements its own with help of
webrtc::LoudnessHistogram.
2. Doesn't forget its past at every GetRmsErrorDb call.
2. Makes AgcManagerDirect use AdaptiveModeLevelEstimatorAgc instead of
webrtc::Agc if the use_agc2_level_estimation flag is set.
Bug: webrtc:7494
Change-Id: I8df3f52e322d433eb5ce5297f4236af2f1877b04
Reviewed-on: https://webrtc-review.googlesource.com/86603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23875}
87 lines
3.4 KiB
C++
87 lines
3.4 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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#include <cmath>
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namespace webrtc {
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constexpr float kMinFloatS16Value = -32768.f;
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constexpr float kMaxFloatS16Value = 32767.f;
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constexpr float kMaxAbsFloatS16Value = 32768.0f;
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constexpr size_t kFrameDurationMs = 10;
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constexpr size_t kSubFramesInFrame = 20;
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constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
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constexpr float kAttackFilterConstant = 0.f;
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// Adaptive digital gain applier settings below.
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constexpr float kMaxGainChangePerSecondDb = 3.f;
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constexpr float kMaxGainChangePerFrameDb =
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kMaxGainChangePerSecondDb * kFrameDurationMs / 1000.f;
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constexpr float kHeadroomDbfs = 1.f;
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constexpr float kMaxGainDb = 30.f;
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constexpr float kInitialAdaptiveDigitalGainDb = 8.f;
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// This parameter must be tuned together with the noise estimator.
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constexpr float kMaxNoiseLevelDbfs = -50.f;
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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 = 0.4f;
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// The amount of 'memory' of the Level Estimator. Decides leak factors.
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constexpr size_t kFullBufferSizeMs = 1600;
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constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
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constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
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// Saturation Protector settings.
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constexpr float kInitialSaturationMarginDb = 17.f;
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constexpr size_t 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 size_t 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 size_t kInterpolatedGainCurveKneePoints = 22;
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constexpr size_t kInterpolatedGainCurveBeyondKneePoints = 10;
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constexpr size_t 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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