/* * 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_ANALOG_AGC_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_ #include "typedefs.h" #include "gain_control.h" #include "digital_agc.h" //#define AGC_DEBUG //#define MIC_LEVEL_FEEDBACK #ifdef AGC_DEBUG #include #endif /* Analog Automatic Gain Control variables: * Constant declarations (inner limits inside which no changes are done) * In the beginning the range is narrower to widen as soon as the measure * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0 * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm * The limits are created by running the AGC with a file having the desired * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined * by out=10*log10(in/260537279.7); Set the target level to the average level * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) ) */ #define RXX_BUFFER_LEN 10 static const WebRtc_Word16 kMsecSpeechInner = 520; static const WebRtc_Word16 kMsecSpeechOuter = 340; static const WebRtc_Word16 kNormalVadThreshold = 400; static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156 static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977 typedef struct { // Configurable parameters/variables WebRtc_UWord32 fs; // Sampling frequency WebRtc_Word16 compressionGaindB; // Fixed gain level in dB WebRtc_Word16 targetLevelDbfs; // Target level in -dBfs of envelope (default -3) WebRtc_Word16 agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig) WebRtc_UWord8 limiterEnable; // Enabling limiter (on/off (default off)) WebRtcAgc_config_t defaultConfig; WebRtcAgc_config_t usedConfig; // General variables WebRtc_Word16 initFlag; WebRtc_Word16 lastError; // Target level parameters // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7) WebRtc_Word32 analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs WebRtc_Word32 startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs WebRtc_Word32 startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs WebRtc_Word32 upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs WebRtc_Word32 lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs WebRtc_Word32 upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs WebRtc_Word32 lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs WebRtc_UWord16 targetIdx; // Table index for corresponding target level #ifdef MIC_LEVEL_FEEDBACK WebRtc_UWord16 targetIdxOffset; // Table index offset for level compensation #endif WebRtc_Word16 analogTarget; // Digital reference level in ENV scale // Analog AGC specific variables WebRtc_Word32 filterState[8]; // For downsampling wb to nb WebRtc_Word32 upperLimit; // Upper limit for mic energy WebRtc_Word32 lowerLimit; // Lower limit for mic energy WebRtc_Word32 Rxx160w32; // Average energy for one frame WebRtc_Word32 Rxx16_LPw32; // Low pass filtered subframe energies WebRtc_Word32 Rxx160_LPw32; // Low pass filtered frame energies WebRtc_Word32 Rxx16_LPw32Max; // Keeps track of largest energy subframe WebRtc_Word32 Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies WebRtc_Word32 Rxx16w32_array[2][5];// Energy values of microphone signal WebRtc_Word32 env[2][10]; // Envelope values of subframes WebRtc_Word16 Rxx16pos; // Current position in the Rxx16_vectorw32 WebRtc_Word16 envSum; // Filtered scaled envelope in subframes WebRtc_Word16 vadThreshold; // Threshold for VAD decision WebRtc_Word16 inActive; // Inactive time in milliseconds WebRtc_Word16 msTooLow; // Milliseconds of speech at a too low level WebRtc_Word16 msTooHigh; // Milliseconds of speech at a too high level WebRtc_Word16 changeToSlowMode; // Change to slow mode after some time at target WebRtc_Word16 firstCall; // First call to the process-function WebRtc_Word16 msZero; // Milliseconds of zero input WebRtc_Word16 msecSpeechOuterChange;// Min ms of speech between volume changes WebRtc_Word16 msecSpeechInnerChange;// Min ms of speech between volume changes WebRtc_Word16 activeSpeech; // Milliseconds of active speech WebRtc_Word16 muteGuardMs; // Counter to prevent mute action WebRtc_Word16 inQueue; // 10 ms batch indicator // Microphone level variables WebRtc_Word32 micRef; // Remember ref. mic level for virtual mic WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly WebRtc_Word32 micVol; // Remember volume between frames WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain WebRtc_Word32 maxAnalog; // Maximum possible analog volume level WebRtc_Word32 maxInit; // Initial value of "max" WebRtc_Word32 minLevel; // Minimum possible volume level WebRtc_Word32 minOutput; // Minimum output volume level WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input WebRtc_Word16 scale; // Scale factor for internal volume levels #ifdef MIC_LEVEL_FEEDBACK WebRtc_Word16 numBlocksMicLvlSat; WebRtc_UWord8 micLvlSat; #endif // Structs for VAD and digital_agc AgcVad_t vadMic; DigitalAgc_t digitalAgc; #ifdef AGC_DEBUG FILE* fpt; FILE* agcLog; WebRtc_Word32 fcount; #endif WebRtc_Word16 lowLevelSignal; } Agc_t; #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_