Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Moved src/ to webrtc/ to easily diff against the third_party/webrtc in the chromium tree * ARM/NEON/MIPS support is not yet hooked up * Tests have not been copied
77 lines
2.9 KiB
C
77 lines
2.9 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 "typedefs.h"
|
|
#include "signal_processing_library.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
|
|
{
|
|
WebRtc_Word32 downState[8];
|
|
WebRtc_Word16 HPstate;
|
|
WebRtc_Word16 counter;
|
|
WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
|
|
WebRtc_Word16 meanLongTerm; // Q10
|
|
WebRtc_Word32 varianceLongTerm; // Q8
|
|
WebRtc_Word16 stdLongTerm; // Q10
|
|
WebRtc_Word16 meanShortTerm; // Q10
|
|
WebRtc_Word32 varianceShortTerm; // Q8
|
|
WebRtc_Word16 stdShortTerm; // Q10
|
|
} AgcVad_t; // total = 54 bytes
|
|
|
|
typedef struct
|
|
{
|
|
WebRtc_Word32 capacitorSlow;
|
|
WebRtc_Word32 capacitorFast;
|
|
WebRtc_Word32 gain;
|
|
WebRtc_Word32 gainTable[32];
|
|
WebRtc_Word16 gatePrevious;
|
|
WebRtc_Word16 agcMode;
|
|
AgcVad_t vadNearend;
|
|
AgcVad_t vadFarend;
|
|
#ifdef AGC_DEBUG
|
|
FILE* logFile;
|
|
int frameCounter;
|
|
#endif
|
|
} DigitalAgc_t;
|
|
|
|
WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
|
|
|
|
WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
|
|
const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
|
|
WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
|
|
WebRtc_Word16 lowLevelSignal);
|
|
|
|
WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
|
|
WebRtc_Word16 nrSamples);
|
|
|
|
void WebRtcAgc_InitVad(AgcVad_t *vadInst);
|
|
|
|
WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
|
|
const WebRtc_Word16 *in, // (i) Speech signal
|
|
WebRtc_Word16 nrSamples); // (i) number of samples
|
|
|
|
WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
|
|
WebRtc_Word16 compressionGaindB, // Q0 (in dB)
|
|
WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
|
|
WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
|