/* * 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 "webrtc/modules/audio_processing/agc/digital_agc.h" #include "webrtc/modules/audio_processing/agc/include/gain_control.h" #include "webrtc/typedefs.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 int16_t kMsecSpeechInner = 520; static const int16_t kMsecSpeechOuter = 340; static const int16_t kNormalVadThreshold = 400; static const int16_t kAlphaShortTerm = 6; // 1 >> 6 = 0.0156 static const int16_t kAlphaLongTerm = 10; // 1 >> 10 = 0.000977 typedef struct { // Configurable parameters/variables uint32_t fs; // Sampling frequency int16_t compressionGaindB; // Fixed gain level in dB int16_t targetLevelDbfs; // Target level in -dBfs of envelope (default -3) int16_t agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig) uint8_t limiterEnable; // Enabling limiter (on/off (default off)) WebRtcAgc_config_t defaultConfig; WebRtcAgc_config_t usedConfig; // General variables int16_t initFlag; int16_t lastError; // Target level parameters // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7) int32_t analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs int32_t startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs int32_t startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs int32_t upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs int32_t lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs int32_t upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs int32_t lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs uint16_t targetIdx; // Table index for corresponding target level #ifdef MIC_LEVEL_FEEDBACK uint16_t targetIdxOffset; // Table index offset for level compensation #endif int16_t analogTarget; // Digital reference level in ENV scale // Analog AGC specific variables int32_t filterState[8]; // For downsampling wb to nb int32_t upperLimit; // Upper limit for mic energy int32_t lowerLimit; // Lower limit for mic energy int32_t Rxx160w32; // Average energy for one frame int32_t Rxx16_LPw32; // Low pass filtered subframe energies int32_t Rxx160_LPw32; // Low pass filtered frame energies int32_t Rxx16_LPw32Max; // Keeps track of largest energy subframe int32_t Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies int32_t Rxx16w32_array[2][5];// Energy values of microphone signal int32_t env[2][10]; // Envelope values of subframes int16_t Rxx16pos; // Current position in the Rxx16_vectorw32 int16_t envSum; // Filtered scaled envelope in subframes int16_t vadThreshold; // Threshold for VAD decision int16_t inActive; // Inactive time in milliseconds int16_t msTooLow; // Milliseconds of speech at a too low level int16_t msTooHigh; // Milliseconds of speech at a too high level int16_t changeToSlowMode; // Change to slow mode after some time at target int16_t firstCall; // First call to the process-function int16_t msZero; // Milliseconds of zero input int16_t msecSpeechOuterChange;// Min ms of speech between volume changes int16_t msecSpeechInnerChange;// Min ms of speech between volume changes int16_t activeSpeech; // Milliseconds of active speech int16_t muteGuardMs; // Counter to prevent mute action int16_t inQueue; // 10 ms batch indicator // Microphone level variables int32_t micRef; // Remember ref. mic level for virtual mic uint16_t gainTableIdx; // Current position in virtual gain table int32_t micGainIdx; // Gain index of mic level to increase slowly int32_t micVol; // Remember volume between frames int32_t maxLevel; // Max possible vol level, incl dig gain int32_t maxAnalog; // Maximum possible analog volume level int32_t maxInit; // Initial value of "max" int32_t minLevel; // Minimum possible volume level int32_t minOutput; // Minimum output volume level int32_t zeroCtrlMax; // Remember max gain => don't amp low input int32_t lastInMicLevel; int16_t scale; // Scale factor for internal volume levels #ifdef MIC_LEVEL_FEEDBACK int16_t numBlocksMicLvlSat; uint8_t micLvlSat; #endif // Structs for VAD and digital_agc AgcVad_t vadMic; DigitalAgc_t digitalAgc; #ifdef AGC_DEBUG FILE* fpt; FILE* agcLog; int32_t fcount; #endif int16_t lowLevelSignal; } Agc_t; #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_