AnotherFoxGuy/webrtc-audio-processing
1
0
Fork 0
Code Pull Requests Activity

Update to current webrtc library

Browse Source 
This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
Download Patch File Download Diff File Expand all files Collapse all files

95
webrtc/modules/audio_processing/agc/legacy/digital_agc.h
View File

@ -8,58 +8,51 @@
* 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_
#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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/typedefs.h"
#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 ))
namespace webrtc {
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 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
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;
} 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_ComputeDigitalGains(DigitalAgc* digitalAgcInst,
const int16_t* const* inNear,
size_t num_bands,
uint32_t FS,
int16_t lowLevelSignal,
int32_t gains[11]);
int32_t WebRtcAgc_ApplyDigitalGains(const int32_t gains[11],
size_t num_bands,
uint32_t FS,
const int16_t* const* in_near,
int16_t* const* out);
int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
const int16_t* inFar,
@ -67,14 +60,16 @@ int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
void WebRtcAgc_InitVad(AgcVad* vadInst);
int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst, // (i) VAD state
const int16_t* in, // (i) Speech signal
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)
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_
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_