mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-03 15:05:24 +00:00
79 lines
2.8 KiB
C
79 lines
2.8 KiB
C
/*
|
|
* 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_MAIN_SOURCE_DIGITAL_AGC_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
|
|
|
|
#ifdef AGC_DEBUG
|
|
#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_t; // 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_t vadNearend;
|
|
AgcVad_t vadFarend;
|
|
#ifdef AGC_DEBUG
|
|
FILE* logFile;
|
|
int frameCounter;
|
|
#endif
|
|
} DigitalAgc_t;
|
|
|
|
int32_t WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, int16_t agcMode);
|
|
|
|
int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst,
|
|
const int16_t *inNear, const int16_t *inNear_H,
|
|
int16_t *out, int16_t *out_H, uint32_t FS,
|
|
int16_t lowLevelSignal);
|
|
|
|
int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst,
|
|
const int16_t *inFar,
|
|
int16_t nrSamples);
|
|
|
|
void WebRtcAgc_InitVad(AgcVad_t *vadInst);
|
|
|
|
int16_t WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
|
|
const int16_t *in, // (i) Speech signal
|
|
int16_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_MAIN_SOURCE_ANALOG_AGC_H_
|