modules/audio_processing/agc/legacy/digital_agc.h

/*
 *  Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_

#ifdef WEBRTC_AGC_DEBUG_DUMP
#include <stdio.h>
#endif
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/typedefs.h"

// the 32 most significant bits of A(19) * B(26) >> 13
#define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13))
// C + the 32 most significant bits of A * B
#define AGC_SCALEDIFF32(A, B, C) \
  ((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16))

typedef struct {
  int32_t downState[8];
  int16_t HPstate;
  int16_t counter;
  int16_t logRatio;           // log( P(active) / P(inactive) ) (Q10)
  int16_t meanLongTerm;       // Q10
  int32_t varianceLongTerm;   // Q8
  int16_t stdLongTerm;        // Q10
  int16_t meanShortTerm;      // Q10
  int32_t varianceShortTerm;  // Q8
  int16_t stdShortTerm;       // Q10
} AgcVad;                     // total = 54 bytes

typedef struct {
  int32_t capacitorSlow;
  int32_t capacitorFast;
  int32_t gain;
  int32_t gainTable[32];
  int16_t gatePrevious;
  int16_t agcMode;
  AgcVad vadNearend;
  AgcVad vadFarend;
#ifdef WEBRTC_AGC_DEBUG_DUMP
  FILE* logFile;
  int frameCounter;
#endif
} DigitalAgc;

int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);

int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
                                 const int16_t* const* inNear,
                                 size_t num_bands,
                                 int16_t* const* out,
                                 uint32_t FS,
                                 int16_t lowLevelSignal);

int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
                                     const int16_t* inFar,
                                     size_t nrSamples);

void WebRtcAgc_InitVad(AgcVad* vadInst);

int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,    // (i) VAD state
                             const int16_t* in,  // (i) Speech signal
                             size_t nrSamples);  // (i) number of samples

int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,         // Q16
                                     int16_t compressionGaindB,  // Q0 (in dB)
                                     int16_t targetLevelDbfs,    // Q0 (in dB)
                                     uint8_t limiterEnable,
                                     int16_t analogTarget);

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_