webrtc/modules/audio_processing/agc/analog_agc.h

/*
 *  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 <stdio.h>
#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_