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			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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| 
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| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
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| #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
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| 
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| #ifdef AGC_DEBUG
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| #include <stdio.h>
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| #endif
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| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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| #include "webrtc/typedefs.h"
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| 
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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)    ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
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| 
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| typedef struct
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| {
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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_t; // total = 54 bytes
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| 
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| typedef struct
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| {
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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_t      vadNearend;
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|     AgcVad_t      vadFarend;
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| #ifdef AGC_DEBUG
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|     FILE*         logFile;
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|     int           frameCounter;
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| #endif
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| } DigitalAgc_t;
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| 
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| int32_t WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, int16_t agcMode);
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| 
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| int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst,
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|                                  const int16_t *inNear, const int16_t *inNear_H,
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|                                  int16_t *out, int16_t *out_H, uint32_t FS,
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|                                  int16_t lowLevelSignal);
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| 
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| int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst,
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|                                      const int16_t *inFar,
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|                                      int16_t nrSamples);
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| 
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| void WebRtcAgc_InitVad(AgcVad_t *vadInst);
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| 
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| int16_t WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
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|                              const int16_t *in, // (i) Speech signal
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|                              int16_t nrSamples); // (i) number of samples
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| 
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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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| 
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| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
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