| /* |
| * 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 "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_ |