462 lines
17 KiB
C
462 lines
17 KiB
C
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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 MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
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#define 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 <stdint.h> // NOLINT(build/include)
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#include <string.h>
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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 <stdint.h> // NOLINT(build/include)
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#include <assert.h>
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// allpass filter coefficients.
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static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528};
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static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
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typedef struct BufferAgc
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{
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size_t frame_size;
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size_t in_uselen;
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size_t out_uselen;
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int16_t *buffer_in;
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int16_t *buffer_out;
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}BufferAgc;
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typedef struct VadEnergy
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{
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size_t sil2spe; // silence to speech
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size_t spe2sil; // speech to silence
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size_t amth; // threshold for energy based vad
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size_t VAD; // 0(silence) or 1(speech)
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size_t specnt; // count for speech frame
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size_t silcnt; // count for silence frame
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size_t silforReset; // reset activeSpeech and Rxx16_LPw32Max
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size_t BeforeSilEnd; // frames are needed to enter the silent segment
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size_t lastSilcent;
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}VadEnergy;
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typedef struct {
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int32_t downState[8];
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int16_t HPstate;
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int16_t counter;
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int16_t logRatio; // log( P(active) / P(inactive) ) (Q10)
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int16_t meanLongTerm; // Q10
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int32_t varianceLongTerm; // Q8
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int16_t stdLongTerm; // Q10
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int16_t meanShortTerm; // Q10
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int32_t varianceShortTerm; // Q8
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int16_t stdShortTerm; // Q10
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} AgcVad; // total = 54 bytes
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typedef struct {
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int32_t capacitorSlow;
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int32_t capacitorFast;
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int32_t gain;
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int32_t gainTable[32];
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int16_t gatePrevious;
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int16_t agcMode;
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AgcVad vadNearend;
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AgcVad vadFarend;
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#ifdef WEBRTC_AGC_DEBUG_DUMP
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FILE* logFile;
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int frameCounter;
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#endif
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} DigitalAgc;
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void buffer_agcProcess_init(BufferAgc *hd);
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void buffer_agcProcess_deinit(void *hd);
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void vad_init (VadEnergy *state);
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int32_t WebRtcAgc_InitDigital(DigitalAgc *digitalAgcInst, int16_t agcMode);
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int32_t WebRtcAgc_ProcessDigital(DigitalAgc *digitalAgcInst,
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int16_t *const *inNear,
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size_t num_bands,
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int16_t *const *out,
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uint32_t FS,
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int16_t lowLevelSignal);
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int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc *digitalAgcInst,
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const int16_t *inFar,
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size_t nrSamples);
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void WebRtcAgc_InitVad(AgcVad *vadInst);
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int16_t WebRtcAgc_ProcessVad(AgcVad *vadInst, // (i) VAD state
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const int16_t *in, // (i) Speech signal
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size_t nrSamples); // (i) number of samples
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int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
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int16_t compressionGaindB, // Q0 (in dB)
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int16_t targetLevelDbfs, // Q0 (in dB)
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uint8_t limiterEnable,
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int16_t analogTarget);
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// Errors
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#define AGC_UNSPECIFIED_ERROR 18000
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#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
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#define AGC_UNINITIALIZED_ERROR 18002
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#define AGC_NULL_POINTER_ERROR 18003
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#define AGC_BAD_PARAMETER_ERROR 18004
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// Warnings
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#define AGC_BAD_PARAMETER_WARNING 18050
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enum {
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kAgcModeUnchanged,
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kAgcModeAdaptiveAnalog,
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kAgcModeAdaptiveDigital,
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kAgcModeFixedDigital
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};
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enum {
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kAgcFalse = 0, kAgcTrue
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};
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typedef struct {
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int16_t targetLevelDbfs; // default 3 (-3 dBOv)
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int16_t compressionGaindB; // default 9 dB
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uint8_t limiterEnable; // default kAgcTrue (on)
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} WebRtcAgcConfig;
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#if defined(__cplusplus)
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extern "C" {
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#endif
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/*
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* This function analyses the number of samples passed to
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* farend and produces any error code that could arise.
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*
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* Input:
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* - agcInst : AGC instance.
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* - samples : Number of samples in input vector.
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error.
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*/
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int WebRtcAgc_GetAddFarendError(void *state, size_t samples);
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/*
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* This function processes a 10 ms frame of far-end speech to determine
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* if there is active speech. The length of the input speech vector must be
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* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
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* FS=48000).
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*
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* Input:
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* - agcInst : AGC instance.
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* - inFar : Far-end input speech vector
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* - samples : Number of samples in input vector
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_AddFarend(void *agcInst, const int16_t *inFar, size_t samples);
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/*
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* This function processes a 10 ms frame of microphone speech to determine
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* if there is active speech. The length of the input speech vector must be
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* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
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* FS=48000). For very low input levels, the input signal is increased in level
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* by multiplying and overwriting the samples in inMic[].
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*
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* This function should be called before any further processing of the
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* near-end microphone signal.
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*
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* Input:
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* - agcInst : AGC instance.
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* - inMic : Microphone input speech vector for each band
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* - num_bands : Number of bands in input vector
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* - samples : Number of samples in input vector
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_AddMic(void *agcInst,
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int16_t *const *inMic,
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size_t num_bands,
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size_t samples);
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/*
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* This function replaces the analog microphone with a virtual one.
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* It is a digital gain applied to the input signal and is used in the
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* agcAdaptiveDigital mode where no microphone level is adjustable. The length
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* of the input speech vector must be given in samples (80 when FS=8000, and 160
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* when FS=16000, FS=32000 or FS=48000).
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*
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* Input:
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* - agcInst : AGC instance.
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* - inMic : Microphone input speech vector for each band
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* - num_bands : Number of bands in input vector
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* - samples : Number of samples in input vector
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* - micLevelIn : Input level of microphone (static)
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*
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* Output:
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* - inMic : Microphone output after processing (L band)
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* - inMic_H : Microphone output after processing (H band)
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* - micLevelOut : Adjusted microphone level after processing
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_VirtualMic(void *agcInst,
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int16_t *const *inMic,
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size_t num_bands,
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size_t samples,
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int32_t micLevelIn,
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int32_t *micLevelOut);
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/*
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* This function processes a 10 ms frame and adjusts (normalizes) the gain both
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* analog and digitally. The gain adjustments are done only during active
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* periods of speech. The length of the speech vectors must be given in samples
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* (80 when FS=8000, and 160 when FS=16000, FS=32000 or FS=48000). The echo
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* parameter can be used to ensure the AGC will not adjust upward in the
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* presence of echo.
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*
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* This function should be called after processing the near-end microphone
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* signal, in any case after any echo cancellation.
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*
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* Input:
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* - agcInst : AGC instance
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* - inNear : Near-end input speech vector for each band
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* - num_bands : Number of bands in input/output vector
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* - samples : Number of samples in input/output vector
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* - inMicLevel : Current microphone volume level
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* - echo : Set to 0 if the signal passed to add_mic is
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* almost certainly free of echo; otherwise set
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* to 1. If you have no information regarding echo
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* set to 0.
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*
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* Output:
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* - outMicLevel : Adjusted microphone volume level
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* - out : Gain-adjusted near-end speech vector
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* : May be the same vector as the input.
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* - saturationWarning : A returned value of 1 indicates a saturation event
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* has occurred and the volume cannot be further
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* reduced. Otherwise will be set to 0.
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_Process(void *agcInst,
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int16_t *const *inNear,
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size_t num_bands,
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size_t samples,
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int16_t *const *out,
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int32_t inMicLevel,
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int32_t *outMicLevel,
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int16_t echo,
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uint8_t *saturationWarning);
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/*
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* This function sets the config parameters (targetLevelDbfs,
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* compressionGaindB and limiterEnable).
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*
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* Input:
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* - agcInst : AGC instance
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* - config : config struct
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*
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* Output:
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_set_config(void *agcInst, WebRtcAgcConfig config);
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/*
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* This function returns the config parameters (targetLevelDbfs,
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* compressionGaindB and limiterEnable).
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*
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* Input:
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* - agcInst : AGC instance
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*
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* Output:
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* - config : config struct
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*
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* Return value:
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* : 0 - Normal operation.
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* : -1 - Error
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*/
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int WebRtcAgc_get_config(void *agcInst, WebRtcAgcConfig *config);
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/*
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* This function creates and returns an AGC instance, which will contain the
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* state information for one (duplex) channel.
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*/
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void *WebRtcAgc_Create(void);
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/*
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* This function frees the AGC instance created at the beginning.
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*
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* Input:
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* - agcInst : AGC instance.
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*/
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void WebRtcAgc_Free(void *agcInst);
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/*
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* This function initializes an AGC instance.
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*
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* Input:
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* - agcInst : AGC instance.
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* - minLevel : Minimum possible mic level
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* - maxLevel : Maximum possible mic level
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* - agcMode : 0 - Unchanged
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* : 1 - Adaptive Analog Automatic Gain Control -3dBOv
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* : 2 - Adaptive Digital Automatic Gain Control -3dBOv
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* : 3 - Fixed Digital Gain 0dB
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* - fs : Sampling frequency
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int WebRtcAgc_Init(void *agcInst,
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int32_t minLevel,
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int32_t maxLevel,
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int16_t agcMode,
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uint32_t fs);
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#if defined(__cplusplus)
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}
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#endif
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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 = 190; // 520
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static const int16_t kMsecSpeechOuter = 120; // 340
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static const int16_t kNormalVadThreshold = 400;
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static const int16_t kAlphaShortTerm = 4; // 1 >> 6 = 0.0156
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static const int16_t kAlphaLongTerm = 8; // 1 >> 10 = 0.000977
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typedef struct {
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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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|
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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 msChangeLarge;
|
||
|
|
int32_t silEnd_micVol; // volume when silent is (spe2sil-10)
|
||
|
|
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 vadMic;
|
||
|
|
DigitalAgc digitalAgc;
|
||
|
|
// AGC process buffer
|
||
|
|
BufferAgc bufferAgc;
|
||
|
|
// Energy VAD
|
||
|
|
VadEnergy vad_energy;
|
||
|
|
#ifdef WEBRTC_AGC_DEBUG_DUMP
|
||
|
|
FILE* fpt;
|
||
|
|
FILE* agcLog;
|
||
|
|
int32_t fcount;
|
||
|
|
#endif
|
||
|
|
|
||
|
|
int16_t lowLevelSignal;
|
||
|
|
} LegacyAgc;
|
||
|
|
|
||
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
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