mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
79 lines
2.8 KiB
C
79 lines
2.8 KiB
C
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
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#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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#include <stdio.h>
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#endif
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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// the 32 most significant bits of A(19) * B(26) >> 13
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#define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13))
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// C + the 32 most significant bits of A * B
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#define AGC_SCALEDIFF32(A, B, C) \
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((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16))
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typedef struct {
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int32_t downState[8];
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int16_t HPstate;
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int16_t counter;
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int16_t logRatio; // log( P(active) / P(inactive) ) (Q10)
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int16_t meanLongTerm; // Q10
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int32_t varianceLongTerm; // Q8
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int16_t stdLongTerm; // Q10
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int16_t meanShortTerm; // Q10
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int32_t varianceShortTerm; // Q8
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int16_t stdShortTerm; // Q10
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} AgcVad; // total = 54 bytes
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typedef struct {
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int32_t capacitorSlow;
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int32_t capacitorFast;
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int32_t gain;
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int32_t gainTable[32];
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int16_t gatePrevious;
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int16_t agcMode;
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AgcVad vadNearend;
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AgcVad vadFarend;
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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FILE* logFile;
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int frameCounter;
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#endif
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} DigitalAgc;
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int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
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int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
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const int16_t* const* inNear,
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size_t num_bands,
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int16_t* const* out,
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uint32_t FS,
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int16_t lowLevelSignal);
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int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
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const int16_t* inFar,
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size_t nrSamples);
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void WebRtcAgc_InitVad(AgcVad* vadInst);
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int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst, // (i) VAD state
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const int16_t* in, // (i) Speech signal
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size_t nrSamples); // (i) number of samples
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int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable, // Q16
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int16_t compressionGaindB, // Q0 (in dB)
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int16_t targetLevelDbfs, // Q0 (in dB)
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uint8_t limiterEnable,
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int16_t analogTarget);
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#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
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