/* * 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 #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_