| /* |
| * 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_ |
