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Diffstat (limited to 'src/modules/audio_processing/ns/main/source/ns_core.h')
| -rw-r--r-- | src/modules/audio_processing/ns/main/source/ns_core.h | 179 |
1 files changed, 179 insertions, 0 deletions
diff --git a/src/modules/audio_processing/ns/main/source/ns_core.h b/src/modules/audio_processing/ns/main/source/ns_core.h new file mode 100644 index 0000000000..f72e22bf1c --- /dev/null +++ b/src/modules/audio_processing/ns/main/source/ns_core.h @@ -0,0 +1,179 @@ +/* + * 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_NS_MAIN_SOURCE_NS_CORE_H_ +#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_ + +#include "defines.h" + +typedef struct NSParaExtract_t_ { + + //bin size of histogram + float binSizeLrt; + float binSizeSpecFlat; + float binSizeSpecDiff; + //range of histogram over which lrt threshold is computed + float rangeAvgHistLrt; + //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain + //thresholds for prior model + float factor1ModelPars; //for lrt and spectral difference + float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech + //peak limit for spectral flatness (varies between 0 and 1) + float thresPosSpecFlat; + //limit on spacing of two highest peaks in histogram: spacing determined by bin size + float limitPeakSpacingSpecFlat; + float limitPeakSpacingSpecDiff; + //limit on relevance of second peak: + float limitPeakWeightsSpecFlat; + float limitPeakWeightsSpecDiff; + //limit on fluctuation of lrt feature + float thresFluctLrt; + //limit on the max and min values for the feature thresholds + float maxLrt; + float minLrt; + float maxSpecFlat; + float minSpecFlat; + float maxSpecDiff; + float minSpecDiff; + //criteria of weight of histogram peak to accept/reject feature + int thresWeightSpecFlat; + int thresWeightSpecDiff; + +} NSParaExtract_t; + +typedef struct NSinst_t_ { + + WebRtc_UWord32 fs; + int blockLen; + int blockLen10ms; + int windShift; + int outLen; + int anaLen; + int magnLen; + int aggrMode; + const float* window; + float dataBuf[ANAL_BLOCKL_MAX]; + float syntBuf[ANAL_BLOCKL_MAX]; + float outBuf[3 * BLOCKL_MAX]; + + int initFlag; + // parameters for quantile noise estimation + float density[SIMULT * HALF_ANAL_BLOCKL]; + float lquantile[SIMULT * HALF_ANAL_BLOCKL]; + float quantile[HALF_ANAL_BLOCKL]; + int counter[SIMULT]; + int updates; + // parameters for Wiener filter + float smooth[HALF_ANAL_BLOCKL]; + float overdrive; + float denoiseBound; + int gainmap; + // fft work arrays. + int ip[IP_LENGTH]; + float wfft[W_LENGTH]; + + // parameters for new method: some not needed, will reduce/cleanup later + WebRtc_Word32 blockInd; //frame index counter + int modelUpdatePars[4]; //parameters for updating or estimating + // thresholds/weights for prior model + float priorModelPars[7]; //parameters for prior model + float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame + float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame + float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing + float priorSpeechProb; //prior speech/noise probability + float featureData[7]; //data for features + float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate + float signalEnergy; //energy of magn + float sumMagn; //sum of magn + float whiteNoiseLevel; //initial noise estimate + float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate + float pinkNoiseNumerator; //pink noise parameter: numerator + float pinkNoiseExp; //pink noise parameter: power of freq + NSParaExtract_t featureExtractionParams; //parameters for feature extraction + //histograms for parameter estimation + int histLrt[HIST_PAR_EST]; + int histSpecFlat[HIST_PAR_EST]; + int histSpecDiff[HIST_PAR_EST]; + //quantities for high band estimate + float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT + float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB + +} NSinst_t; + + +#ifdef __cplusplus +extern "C" { +#endif + +/**************************************************************************** + * WebRtcNs_InitCore(...) + * + * This function initializes a noise suppression instance + * + * Input: + * - inst : Instance that should be initialized + * - fs : Sampling frequency + * + * Output: + * - inst : Initialized instance + * + * Return value : 0 - Ok + * -1 - Error + */ +int WebRtcNs_InitCore(NSinst_t *inst, WebRtc_UWord32 fs); + +/**************************************************************************** + * WebRtcNs_set_policy_core(...) + * + * This changes the aggressiveness of the noise suppression method. + * + * Input: + * - inst : Instance that should be initialized + * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB) + * + * Output: + * - NS_inst : Initialized instance + * + * Return value : 0 - Ok + * -1 - Error + */ +int WebRtcNs_set_policy_core(NSinst_t *inst, int mode); + +/**************************************************************************** + * WebRtcNs_ProcessCore + * + * Do noise suppression. + * + * Input: + * - inst : Instance that should be initialized + * - inFrameLow : Input speech frame for lower band + * - inFrameHigh : Input speech frame for higher band + * + * Output: + * - inst : Updated instance + * - outFrameLow : Output speech frame for lower band + * - outFrameHigh : Output speech frame for higher band + * + * Return value : 0 - OK + * -1 - Error + */ + + +int WebRtcNs_ProcessCore(NSinst_t *inst, + short *inFrameLow, + short *inFrameHigh, + short *outFrameLow, + short *outFrameHigh); + + +#ifdef __cplusplus +} +#endif +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_ |