mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
a12c42a6b2
Usage replaced with stdint.h, rtc_base/system/arch.h and rtc_base/system/unused.h, as appropriate. Bug: webrtc:6854 Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18 Reviewed-on: https://webrtc-review.googlesource.com/90249 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24100}
78 lines
2.8 KiB
C
78 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 MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
|
|
|
|
#ifdef WEBRTC_AGC_DEBUG_DUMP
|
|
#include <stdio.h>
|
|
#endif
|
|
#include "common_audio/signal_processing/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 {
|
|
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 // MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
|