1 /* 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_ 13 14 #include "typedefs.h" 15 #include "gain_control.h" 16 #include "digital_agc.h" 17 18 //#define AGC_DEBUG 19 //#define MIC_LEVEL_FEEDBACK 20 #ifdef AGC_DEBUG 21 #include <stdio.h> 22 #endif 23 24 /* Analog Automatic Gain Control variables: 25 * Constant declarations (inner limits inside which no changes are done) 26 * In the beginning the range is narrower to widen as soon as the measure 27 * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0 28 * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal 29 * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm 30 * The limits are created by running the AGC with a file having the desired 31 * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined 32 * by out=10*log10(in/260537279.7); Set the target level to the average level 33 * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in 34 * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) ) 35 */ 36 #define RXX_BUFFER_LEN 10 37 38 static const WebRtc_Word16 kMsecSpeechInner = 520; 39 static const WebRtc_Word16 kMsecSpeechOuter = 340; 40 41 static const WebRtc_Word16 kNormalVadThreshold = 400; 42 43 static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156 44 static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977 45 46 typedef struct 47 { 48 // Configurable parameters/variables 49 WebRtc_UWord32 fs; // Sampling frequency 50 WebRtc_Word16 compressionGaindB; // Fixed gain level in dB 51 WebRtc_Word16 targetLevelDbfs; // Target level in -dBfs of envelope (default -3) 52 WebRtc_Word16 agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig) 53 WebRtc_UWord8 limiterEnable; // Enabling limiter (on/off (default off)) 54 WebRtcAgc_config_t defaultConfig; 55 WebRtcAgc_config_t usedConfig; 56 57 // General variables 58 WebRtc_Word16 initFlag; 59 WebRtc_Word16 lastError; 60 61 // Target level parameters 62 // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7) 63 WebRtc_Word32 analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs 64 WebRtc_Word32 startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs 65 WebRtc_Word32 startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs 66 WebRtc_Word32 upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs 67 WebRtc_Word32 lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs 68 WebRtc_Word32 upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs 69 WebRtc_Word32 lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs 70 WebRtc_UWord16 targetIdx; // Table index for corresponding target level 71 #ifdef MIC_LEVEL_FEEDBACK 72 WebRtc_UWord16 targetIdxOffset; // Table index offset for level compensation 73 #endif 74 WebRtc_Word16 analogTarget; // Digital reference level in ENV scale 75 76 // Analog AGC specific variables 77 WebRtc_Word32 filterState[8]; // For downsampling wb to nb 78 WebRtc_Word32 upperLimit; // Upper limit for mic energy 79 WebRtc_Word32 lowerLimit; // Lower limit for mic energy 80 WebRtc_Word32 Rxx160w32; // Average energy for one frame 81 WebRtc_Word32 Rxx16_LPw32; // Low pass filtered subframe energies 82 WebRtc_Word32 Rxx160_LPw32; // Low pass filtered frame energies 83 WebRtc_Word32 Rxx16_LPw32Max; // Keeps track of largest energy subframe 84 WebRtc_Word32 Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies 85 WebRtc_Word32 Rxx16w32_array[2][5];// Energy values of microphone signal 86 WebRtc_Word32 env[2][10]; // Envelope values of subframes 87 88 WebRtc_Word16 Rxx16pos; // Current position in the Rxx16_vectorw32 89 WebRtc_Word16 envSum; // Filtered scaled envelope in subframes 90 WebRtc_Word16 vadThreshold; // Threshold for VAD decision 91 WebRtc_Word16 inActive; // Inactive time in milliseconds 92 WebRtc_Word16 msTooLow; // Milliseconds of speech at a too low level 93 WebRtc_Word16 msTooHigh; // Milliseconds of speech at a too high level 94 WebRtc_Word16 changeToSlowMode; // Change to slow mode after some time at target 95 WebRtc_Word16 firstCall; // First call to the process-function 96 WebRtc_Word16 msZero; // Milliseconds of zero input 97 WebRtc_Word16 msecSpeechOuterChange;// Min ms of speech between volume changes 98 WebRtc_Word16 msecSpeechInnerChange;// Min ms of speech between volume changes 99 WebRtc_Word16 activeSpeech; // Milliseconds of active speech 100 WebRtc_Word16 muteGuardMs; // Counter to prevent mute action 101 WebRtc_Word16 inQueue; // 10 ms batch indicator 102 103 // Microphone level variables 104 WebRtc_Word32 micRef; // Remember ref. mic level for virtual mic 105 WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table 106 WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly 107 WebRtc_Word32 micVol; // Remember volume between frames 108 WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain 109 WebRtc_Word32 maxAnalog; // Maximum possible analog volume level 110 WebRtc_Word32 maxInit; // Initial value of "max" 111 WebRtc_Word32 minLevel; // Minimum possible volume level 112 WebRtc_Word32 minOutput; // Minimum output volume level 113 WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input 114 115 WebRtc_Word16 scale; // Scale factor for internal volume levels 116 #ifdef MIC_LEVEL_FEEDBACK 117 WebRtc_Word16 numBlocksMicLvlSat; 118 WebRtc_UWord8 micLvlSat; 119 #endif 120 // Structs for VAD and digital_agc 121 AgcVad_t vadMic; 122 DigitalAgc_t digitalAgc; 123 124 #ifdef AGC_DEBUG 125 FILE* fpt; 126 FILE* agcLog; 127 WebRtc_Word32 fcount; 128 #endif 129 130 WebRtc_Word16 lowLevelSignal; 131 } Agc_t; 132 133 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_ 134