Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
This commit is contained in:
Arun Raghavan
133
webrtc/modules/audio_processing/agc/legacy/analog_agc.h
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133
webrtc/modules/audio_processing/agc/legacy/analog_agc.h
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
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//#define MIC_LEVEL_FEEDBACK
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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#include <stdio.h>
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#endif
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#include "webrtc/modules/audio_processing/agc/legacy/digital_agc.h"
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#include "webrtc/modules/audio_processing/agc/legacy/gain_control.h"
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#include "webrtc/typedefs.h"
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/* Analog Automatic Gain Control variables:
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* Constant declarations (inner limits inside which no changes are done)
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* In the beginning the range is narrower to widen as soon as the measure
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* 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
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* and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
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* go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
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* The limits are created by running the AGC with a file having the desired
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* signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
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* by out=10*log10(in/260537279.7); Set the target level to the average level
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* of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
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* Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
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*/
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#define RXX_BUFFER_LEN 10
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static const int16_t kMsecSpeechInner = 520;
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static const int16_t kMsecSpeechOuter = 340;
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static const int16_t kNormalVadThreshold = 400;
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static const int16_t kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
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static const int16_t kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
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typedef struct
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{
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// Configurable parameters/variables
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uint32_t fs; // Sampling frequency
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int16_t compressionGaindB; // Fixed gain level in dB
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int16_t targetLevelDbfs; // Target level in -dBfs of envelope (default -3)
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int16_t agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig)
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uint8_t limiterEnable; // Enabling limiter (on/off (default off))
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WebRtcAgcConfig defaultConfig;
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WebRtcAgcConfig usedConfig;
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// General variables
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int16_t initFlag;
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int16_t lastError;
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// Target level parameters
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// Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
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int32_t analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs
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int32_t startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs
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int32_t startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs
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int32_t upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs
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int32_t lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs
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int32_t upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs
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int32_t lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs
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uint16_t targetIdx; // Table index for corresponding target level
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#ifdef MIC_LEVEL_FEEDBACK
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uint16_t targetIdxOffset; // Table index offset for level compensation
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#endif
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int16_t analogTarget; // Digital reference level in ENV scale
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// Analog AGC specific variables
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int32_t filterState[8]; // For downsampling wb to nb
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int32_t upperLimit; // Upper limit for mic energy
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int32_t lowerLimit; // Lower limit for mic energy
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int32_t Rxx160w32; // Average energy for one frame
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int32_t Rxx16_LPw32; // Low pass filtered subframe energies
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int32_t Rxx160_LPw32; // Low pass filtered frame energies
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int32_t Rxx16_LPw32Max; // Keeps track of largest energy subframe
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int32_t Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
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int32_t Rxx16w32_array[2][5];// Energy values of microphone signal
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int32_t env[2][10]; // Envelope values of subframes
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int16_t Rxx16pos; // Current position in the Rxx16_vectorw32
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int16_t envSum; // Filtered scaled envelope in subframes
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int16_t vadThreshold; // Threshold for VAD decision
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int16_t inActive; // Inactive time in milliseconds
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int16_t msTooLow; // Milliseconds of speech at a too low level
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int16_t msTooHigh; // Milliseconds of speech at a too high level
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int16_t changeToSlowMode; // Change to slow mode after some time at target
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int16_t firstCall; // First call to the process-function
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int16_t msZero; // Milliseconds of zero input
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int16_t msecSpeechOuterChange;// Min ms of speech between volume changes
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int16_t msecSpeechInnerChange;// Min ms of speech between volume changes
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int16_t activeSpeech; // Milliseconds of active speech
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int16_t muteGuardMs; // Counter to prevent mute action
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int16_t inQueue; // 10 ms batch indicator
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// Microphone level variables
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int32_t micRef; // Remember ref. mic level for virtual mic
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uint16_t gainTableIdx; // Current position in virtual gain table
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int32_t micGainIdx; // Gain index of mic level to increase slowly
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int32_t micVol; // Remember volume between frames
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int32_t maxLevel; // Max possible vol level, incl dig gain
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int32_t maxAnalog; // Maximum possible analog volume level
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int32_t maxInit; // Initial value of "max"
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int32_t minLevel; // Minimum possible volume level
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int32_t minOutput; // Minimum output volume level
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int32_t zeroCtrlMax; // Remember max gain => don't amp low input
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int32_t lastInMicLevel;
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int16_t scale; // Scale factor for internal volume levels
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#ifdef MIC_LEVEL_FEEDBACK
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int16_t numBlocksMicLvlSat;
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uint8_t micLvlSat;
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#endif
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// Structs for VAD and digital_agc
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AgcVad vadMic;
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DigitalAgc digitalAgc;
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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FILE* fpt;
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FILE* agcLog;
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int32_t fcount;
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#endif
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int16_t lowLevelSignal;
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} LegacyAgc;
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
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