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/*
* 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.
*/
/*
* Specifies the interface for the AEC core.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
#include <stdio.h>
#include "signal_processing_library.h"
#include "typedefs.h"
#define FRAME_LEN 80
#define PART_LEN 64 // Length of partition
#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
#define NR_PART 12 // Number of partitions in filter.
#define PREF_BAND_SIZE 24
// Delay estimator constants, used for logging.
enum { kMaxDelayBlocks = 60 };
enum { kLookaheadBlocks = 15 };
enum { kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks };
typedef float complex_t[2];
// For performance reasons, some arrays of complex numbers are replaced by twice
// as long arrays of float, all the real parts followed by all the imaginary
// ones (complex_t[SIZE] -> float[2][SIZE]). This allows SIMD optimizations and
// is better than two arrays (one for the real parts and one for the imaginary
// parts) as this other way would require two pointers instead of one and cause
// extra register spilling. This also allows the offsets to be calculated at
// compile time.
// Metrics
enum {offsetLevel = -100};
typedef struct {
float sfrsum;
int sfrcounter;
float framelevel;
float frsum;
int frcounter;
float minlevel;
float averagelevel;
} power_level_t;
typedef struct {
float instant;
float average;
float min;
float max;
float sum;
float hisum;
float himean;
int counter;
int hicounter;
} stats_t;
typedef struct {
int farBufWritePos, farBufReadPos;
int knownDelay;
int inSamples, outSamples;
int delayEstCtr;
void *nearFrBuf, *outFrBuf;
void *nearFrBufH;
void *outFrBufH;
float dBuf[PART_LEN2]; // nearend
float eBuf[PART_LEN2]; // error
float dBufH[PART_LEN2]; // nearend
float xPow[PART_LEN1];
float dPow[PART_LEN1];
float dMinPow[PART_LEN1];
float dInitMinPow[PART_LEN1];
float *noisePow;
float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
complex_t sde[PART_LEN1]; // cross-psd of nearend and error
complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
float hNs[PART_LEN1];
float hNlFbMin, hNlFbLocalMin;
float hNlXdAvgMin;
int hNlNewMin, hNlMinCtr;
float overDrive, overDriveSm;
float targetSupp, minOverDrive;
float outBuf[PART_LEN];
int delayIdx;
short stNearState, echoState;
short divergeState;
int xfBufBlockPos;
void* far_buf;
void* far_buf_windowed;
int system_delay; // Current system delay buffered in AEC.
int mult; // sampling frequency multiple
int sampFreq;
WebRtc_UWord32 seed;
float mu; // stepsize
float errThresh; // error threshold
int noiseEstCtr;
power_level_t farlevel;
power_level_t nearlevel;
power_level_t linoutlevel;
power_level_t nlpoutlevel;
int metricsMode;
int stateCounter;
stats_t erl;
stats_t erle;
stats_t aNlp;
stats_t rerl;
// Quantities to control H band scaling for SWB input
int freq_avg_ic; //initial bin for averaging nlp gain
int flag_Hband_cn; //for comfort noise
float cn_scale_Hband; //scale for comfort noise in H band
int delay_histogram[kHistorySizeBlocks];
int delay_logging_enabled;
void* delay_estimator;
#ifdef WEBRTC_AEC_DEBUG_DUMP
void* far_time_buf;
FILE *farFile;
FILE *nearFile;
FILE *outFile;
FILE *outLinearFile;
#endif
} aec_t;
typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
typedef void (*WebRtcAec_FilterAdaptation_t)
(aec_t *aec, float *fft, float ef[2][PART_LEN1]);
extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
typedef void (*WebRtcAec_OverdriveAndSuppress_t)
(aec_t *aec, float hNl[PART_LEN1], const float hNlFb, float efw[2][PART_LEN1]);
extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
int WebRtcAec_CreateAec(aec_t **aec);
int WebRtcAec_FreeAec(aec_t *aec);
int WebRtcAec_InitAec(aec_t *aec, int sampFreq);
void WebRtcAec_InitAec_SSE2(void);
void WebRtcAec_InitMetrics(aec_t *aec);
void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend);
void WebRtcAec_ProcessFrame(aec_t* aec,
const short *nearend,
const short *nearendH,
int knownDelay);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
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