| /* |
| * 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 MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_ |
| #define MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_ |
| |
| #include "modules/audio_processing/agc/legacy/digital_agc.h" |
| #include "modules/audio_processing/agc/legacy/gain_control.h" |
| |
| namespace webrtc { |
| |
| /* 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) ) |
| */ |
| constexpr int16_t kRxxBufferLen = 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)) |
| WebRtcAgcConfig defaultConfig; |
| WebRtcAgcConfig 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; // = kRxxBufferLen * 846805; -22 dBfs |
| int32_t startUpperLimit; // = kRxxBufferLen * 1066064; -21 dBfs |
| int32_t startLowerLimit; // = kRxxBufferLen * 672641; -23 dBfs |
| int32_t upperPrimaryLimit; // = kRxxBufferLen * 1342095; -20 dBfs |
| int32_t lowerPrimaryLimit; // = kRxxBufferLen * 534298; -24 dBfs |
| int32_t upperSecondaryLimit; // = kRxxBufferLen * 2677832; -17 dBfs |
| int32_t lowerSecondaryLimit; // = kRxxBufferLen * 267783; -27 dBfs |
| uint16_t targetIdx; // Table index for corresponding target level |
| 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[kRxxBufferLen]; // 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 |
| // Structs for VAD and digital_agc |
| AgcVad vadMic; |
| DigitalAgc digitalAgc; |
| |
| int16_t lowLevelSignal; |
| } LegacyAgc; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_ |
