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_LEGACY_ANALOG_AGC_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_ 13 14 //#define MIC_LEVEL_FEEDBACK 15 #ifdef WEBRTC_AGC_DEBUG_DUMP 16 #include <stdio.h> 17 #endif 18 19 #include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h" 20 #include "webrtc/modules/audio_processing/agc/legacy/gain_control.h" 21 #include "webrtc/typedefs.h" 22 23 /* Analog Automatic Gain Control variables: 24 * Constant declarations (inner limits inside which no changes are done) 25 * In the beginning the range is narrower to widen as soon as the measure 26 * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0 27 * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal 28 * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm 29 * The limits are created by running the AGC with a file having the desired 30 * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined 31 * by out=10*log10(in/260537279.7); Set the target level to the average level 32 * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in 33 * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) ) 34 */ 35 #define RXX_BUFFER_LEN 10 36 37 static const int16_t kMsecSpeechInner = 520; 38 static const int16_t kMsecSpeechOuter = 340; 39 40 static const int16_t kNormalVadThreshold = 400; 41 42 static const int16_t kAlphaShortTerm = 6; // 1 >> 6 = 0.0156 43 static const int16_t kAlphaLongTerm = 10; // 1 >> 10 = 0.000977 44 45 typedef struct 46 { 47 // Configurable parameters/variables 48 uint32_t fs; // Sampling frequency 49 int16_t compressionGaindB; // Fixed gain level in dB 50 int16_t targetLevelDbfs; // Target level in -dBfs of envelope (default -3) 51 int16_t agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig) 52 uint8_t limiterEnable; // Enabling limiter (on/off (default off)) 53 WebRtcAgcConfig defaultConfig; 54 WebRtcAgcConfig usedConfig; 55 56 // General variables 57 int16_t initFlag; 58 int16_t lastError; 59 60 // Target level parameters 61 // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7) 62 int32_t analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs 63 int32_t startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs 64 int32_t startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs 65 int32_t upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs 66 int32_t lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs 67 int32_t upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs 68 int32_t lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs 69 uint16_t targetIdx; // Table index for corresponding target level 70 #ifdef MIC_LEVEL_FEEDBACK 71 uint16_t targetIdxOffset; // Table index offset for level compensation 72 #endif 73 int16_t analogTarget; // Digital reference level in ENV scale 74 75 // Analog AGC specific variables 76 int32_t filterState[8]; // For downsampling wb to nb 77 int32_t upperLimit; // Upper limit for mic energy 78 int32_t lowerLimit; // Lower limit for mic energy 79 int32_t Rxx160w32; // Average energy for one frame 80 int32_t Rxx16_LPw32; // Low pass filtered subframe energies 81 int32_t Rxx160_LPw32; // Low pass filtered frame energies 82 int32_t Rxx16_LPw32Max; // Keeps track of largest energy subframe 83 int32_t Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies 84 int32_t Rxx16w32_array[2][5];// Energy values of microphone signal 85 int32_t env[2][10]; // Envelope values of subframes 86 87 int16_t Rxx16pos; // Current position in the Rxx16_vectorw32 88 int16_t envSum; // Filtered scaled envelope in subframes 89 int16_t vadThreshold; // Threshold for VAD decision 90 int16_t inActive; // Inactive time in milliseconds 91 int16_t msTooLow; // Milliseconds of speech at a too low level 92 int16_t msTooHigh; // Milliseconds of speech at a too high level 93 int16_t changeToSlowMode; // Change to slow mode after some time at target 94 int16_t firstCall; // First call to the process-function 95 int16_t msZero; // Milliseconds of zero input 96 int16_t msecSpeechOuterChange;// Min ms of speech between volume changes 97 int16_t msecSpeechInnerChange;// Min ms of speech between volume changes 98 int16_t activeSpeech; // Milliseconds of active speech 99 int16_t muteGuardMs; // Counter to prevent mute action 100 int16_t inQueue; // 10 ms batch indicator 101 102 // Microphone level variables 103 int32_t micRef; // Remember ref. mic level for virtual mic 104 uint16_t gainTableIdx; // Current position in virtual gain table 105 int32_t micGainIdx; // Gain index of mic level to increase slowly 106 int32_t micVol; // Remember volume between frames 107 int32_t maxLevel; // Max possible vol level, incl dig gain 108 int32_t maxAnalog; // Maximum possible analog volume level 109 int32_t maxInit; // Initial value of "max" 110 int32_t minLevel; // Minimum possible volume level 111 int32_t minOutput; // Minimum output volume level 112 int32_t zeroCtrlMax; // Remember max gain => don't amp low input 113 int32_t lastInMicLevel; 114 115 int16_t scale; // Scale factor for internal volume levels 116 #ifdef MIC_LEVEL_FEEDBACK 117 int16_t numBlocksMicLvlSat; 118 uint8_t micLvlSat; 119 #endif 120 // Structs for VAD and digital_agc 121 AgcVad vadMic; 122 DigitalAgc digitalAgc; 123 124 #ifdef WEBRTC_AGC_DEBUG_DUMP 125 FILE* fpt; 126 FILE* agcLog; 127 int32_t fcount; 128 #endif 129 130 int16_t lowLevelSignal; 131 } LegacyAgc; 132 133 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_ 134
