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Diffstat (limited to 'src/modules/audio_processing/agc/main/source/analog_agc.h')
-rw-r--r-- | src/modules/audio_processing/agc/main/source/analog_agc.h | 133 |
1 files changed, 133 insertions, 0 deletions
diff --git a/src/modules/audio_processing/agc/main/source/analog_agc.h b/src/modules/audio_processing/agc/main/source/analog_agc.h new file mode 100644 index 0000000..b32ac65 --- /dev/null +++ b/src/modules/audio_processing/agc/main/source/analog_agc.h @@ -0,0 +1,133 @@ +/* + * 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 <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 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_ |