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/*
* 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
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_
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