Remove the different block lengths in ns_core
Relanding the CL: https://webrtc-codereview.appspot.com/30539004/
It had to be reverted because some development code was uploaded by mistake.
TBR=bjornv@webrtc.org
BUG=webrtc:3811
Review URL: https://webrtc-codereview.appspot.com/28589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7307 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
index 285e404..5d367ee 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/webrtc/modules/audio_processing/ns/ns_core.c
@@ -90,24 +90,18 @@
if (fs == 8000) {
// We only support 10ms frames
inst->blockLen = 80;
- inst->blockLen10ms = 80;
inst->anaLen = 128;
inst->window = kBlocks80w128;
- inst->outLen = 0;
} else if (fs == 16000) {
// We only support 10ms frames
inst->blockLen = 160;
- inst->blockLen10ms = 160;
inst->anaLen = 256;
inst->window = kBlocks160w256;
- inst->outLen = 0;
} else if (fs == 32000) {
// We only support 10ms frames
inst->blockLen = 160;
- inst->blockLen10ms = 160;
inst->anaLen = 256;
inst->window = kBlocks160w256;
- inst->outLen = 0;
}
inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
@@ -148,13 +142,13 @@
// initialize variables for new method
inst->priorSpeechProb = (float)0.5; // prior prob for speech/noise
for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
- inst->magnPrev[i] = (float)0.0; // previous mag spectrum
+ inst->magnPrev[i] = (float)0.0; // previous mag spectrum
inst->noisePrev[i] = (float)0.0; // previous noise-spectrum
inst->logLrtTimeAvg[i] =
- LRT_FEATURE_THR; // smooth LR ratio (same as threshold)
+ LRT_FEATURE_THR; // smooth LR ratio (same as threshold)
inst->magnAvgPause[i] = (float)0.0; // conservative noise spectrum estimate
- inst->speechProb[i] = (float)0.0; // for estimation of HB in second pass
- inst->initMagnEst[i] = (float)0.0; // initial average mag spectrum
+ inst->speechProb[i] = (float)0.0; // for estimation of HB in second pass
+ inst->initMagnEst[i] = (float)0.0; // initial average mag spectrum
}
// feature quantities
@@ -215,8 +209,6 @@
// default mode
WebRtcNs_set_policy_core(inst, 0);
- memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
-
inst->initFlag = 1;
return 0;
}
@@ -789,250 +781,245 @@
// update analysis buffer for L band
memcpy(inst->analyzeBuf,
- inst->analyzeBuf + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->analyzeBuf + inst->anaLen - inst->blockLen10ms,
+ inst->analyzeBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memcpy(inst->analyzeBuf + inst->anaLen - inst->blockLen,
speechFrame,
- sizeof(float) * inst->blockLen10ms);
+ sizeof(float) * inst->blockLen);
- // check if processing needed
- if (inst->outLen == 0) {
- // windowing
- energy = 0.0;
- for (i = 0; i < inst->anaLen; i++) {
- winData[i] = inst->window[i] * inst->analyzeBuf[i];
- energy += winData[i] * winData[i];
- }
- if (energy == 0.0) {
- // we want to avoid updating statistics in this case:
- // Updating feature statistics when we have zeros only will cause
- // thresholds to move towards zero signal situations. This in turn has the
- // effect that once the signal is "turned on" (non-zero values) everything
- // will be treated as speech and there is no noise suppression effect.
- // Depending on the duration of the inactive signal it takes a
- // considerable amount of time for the system to learn what is noise and
- // what is speech.
- return 0;
- }
+ // windowing
+ energy = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ winData[i] = inst->window[i] * inst->analyzeBuf[i];
+ energy += winData[i] * winData[i];
+ }
+ if (energy == 0.0) {
+ // we want to avoid updating statistics in this case:
+ // Updating feature statistics when we have zeros only will cause
+ // thresholds to move towards zero signal situations. This in turn has the
+ // effect that once the signal is "turned on" (non-zero values) everything
+ // will be treated as speech and there is no noise suppression effect.
+ // Depending on the duration of the inactive signal it takes a
+ // considerable amount of time for the system to learn what is noise and
+ // what is speech.
+ return 0;
+ }
- //
- inst->blockInd++; // Update the block index only when we process a block.
- // FFT
- WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
+ //
+ inst->blockInd++; // Update the block index only when we process a block.
+ // FFT
+ WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
- imag[0] = 0;
- real[0] = winData[0];
- magn[0] = (float)(fabs(real[0]) + 1.0f);
- imag[inst->magnLen - 1] = 0;
- real[inst->magnLen - 1] = winData[1];
- magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
- signalEnergy = (float)(real[0] * real[0]) +
- (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]);
- sumMagn = magn[0] + magn[inst->magnLen - 1];
+ imag[0] = 0;
+ real[0] = winData[0];
+ magn[0] = (float)(fabs(real[0]) + 1.0f);
+ imag[inst->magnLen - 1] = 0;
+ real[inst->magnLen - 1] = winData[1];
+ magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
+ signalEnergy = (float)(real[0] * real[0]) +
+ (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]);
+ sumMagn = magn[0] + magn[inst->magnLen - 1];
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ tmpFloat2 = log((float)(inst->magnLen - 1));
+ sum_log_i = tmpFloat2;
+ sum_log_i_square = tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[inst->magnLen - 1]);
+ sum_log_magn = tmpFloat1;
+ sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
+ }
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ real[i] = winData[2 * i];
+ imag[i] = winData[2 * i + 1];
+ // magnitude spectrum
+ fTmp = real[i] * real[i];
+ fTmp += imag[i] * imag[i];
+ signalEnergy += fTmp;
+ magn[i] = ((float)sqrt(fTmp)) + 1.0f;
+ sumMagn += magn[i];
if (inst->blockInd < END_STARTUP_SHORT) {
- tmpFloat2 = log((float)(inst->magnLen - 1));
- sum_log_i = tmpFloat2;
- sum_log_i_square = tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[inst->magnLen - 1]);
- sum_log_magn = tmpFloat1;
- sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
- }
- for (i = 1; i < inst->magnLen - 1; i++) {
- real[i] = winData[2 * i];
- imag[i] = winData[2 * i + 1];
- // magnitude spectrum
- fTmp = real[i] * real[i];
- fTmp += imag[i] * imag[i];
- signalEnergy += fTmp;
- magn[i] = ((float)sqrt(fTmp)) + 1.0f;
- sumMagn += magn[i];
- if (inst->blockInd < END_STARTUP_SHORT) {
- if (i >= kStartBand) {
- tmpFloat2 = log((float)i);
- sum_log_i += tmpFloat2;
- sum_log_i_square += tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[i]);
- sum_log_magn += tmpFloat1;
- sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
- }
+ if (i >= kStartBand) {
+ tmpFloat2 = log((float)i);
+ sum_log_i += tmpFloat2;
+ sum_log_i_square += tmpFloat2 * tmpFloat2;
+ tmpFloat1 = log(magn[i]);
+ sum_log_magn += tmpFloat1;
+ sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
}
}
- signalEnergy = signalEnergy / ((float)inst->magnLen);
- inst->signalEnergy = signalEnergy;
- inst->sumMagn = sumMagn;
+ }
+ signalEnergy = signalEnergy / ((float)inst->magnLen);
+ inst->signalEnergy = signalEnergy;
+ inst->sumMagn = sumMagn;
- // compute spectral flatness on input spectrum
- WebRtcNs_ComputeSpectralFlatness(inst, magn);
- // quantile noise estimate
- WebRtcNs_NoiseEstimation(inst, magn, noise);
- // compute simplified noise model during startup
- if (inst->blockInd < END_STARTUP_SHORT) {
- // Estimate White noise
- inst->whiteNoiseLevel +=
- sumMagn / ((float)inst->magnLen) * inst->overdrive;
- // Estimate Pink noise parameters
- tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
- tmpFloat1 -= (sum_log_i * sum_log_i);
- tmpFloat2 =
- (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the estimated spectrum to be positive
- if (tmpFloat3 < 0.0f) {
- tmpFloat3 = 0.0f;
- }
- inst->pinkNoiseNumerator += tmpFloat3;
- tmpFloat2 = (sum_log_i * sum_log_magn);
- tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the pink noise power to be in the interval [0, 1];
- if (tmpFloat3 < 0.0f) {
- tmpFloat3 = 0.0f;
- }
- if (tmpFloat3 > 1.0f) {
- tmpFloat3 = 1.0f;
- }
- inst->pinkNoiseExp += tmpFloat3;
+ // compute spectral flatness on input spectrum
+ WebRtcNs_ComputeSpectralFlatness(inst, magn);
+ // quantile noise estimate
+ WebRtcNs_NoiseEstimation(inst, magn, noise);
+ // compute simplified noise model during startup
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ // Estimate White noise
+ inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
+ // Estimate Pink noise parameters
+ tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
+ tmpFloat1 -= (sum_log_i * sum_log_i);
+ tmpFloat2 =
+ (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the estimated spectrum to be positive
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ inst->pinkNoiseNumerator += tmpFloat3;
+ tmpFloat2 = (sum_log_i * sum_log_magn);
+ tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
+ tmpFloat3 = tmpFloat2 / tmpFloat1;
+ // Constrain the pink noise power to be in the interval [0, 1];
+ if (tmpFloat3 < 0.0f) {
+ tmpFloat3 = 0.0f;
+ }
+ if (tmpFloat3 > 1.0f) {
+ tmpFloat3 = 1.0f;
+ }
+ inst->pinkNoiseExp += tmpFloat3;
- // Calculate frequency independent parts of parametric noise estimate.
- if (inst->pinkNoiseExp > 0.0f) {
+ // Calculate frequency independent parts of parametric noise estimate.
+ if (inst->pinkNoiseExp > 0.0f) {
+ // Use pink noise estimate
+ parametric_num =
+ exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
+ parametric_num *= (float)(inst->blockInd + 1);
+ parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
+ }
+ for (i = 0; i < inst->magnLen; i++) {
+ // Estimate the background noise using the white and pink noise
+ // parameters
+ if (inst->pinkNoiseExp == 0.0f) {
+ // Use white noise estimate
+ inst->parametricNoise[i] = inst->whiteNoiseLevel;
+ } else {
// Use pink noise estimate
- parametric_num =
- exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
- parametric_num *= (float)(inst->blockInd + 1);
- parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
+ float use_band = (float)(i < kStartBand ? kStartBand : i);
+ inst->parametricNoise[i] =
+ parametric_num / pow(use_band, parametric_exp);
}
- for (i = 0; i < inst->magnLen; i++) {
- // Estimate the background noise using the white and pink noise
- // parameters
- if (inst->pinkNoiseExp == 0.0f) {
- // Use white noise estimate
- inst->parametricNoise[i] = inst->whiteNoiseLevel;
- } else {
- // Use pink noise estimate
- float use_band = (float)(i < kStartBand ? kStartBand : i);
- inst->parametricNoise[i] =
- parametric_num / pow(use_band, parametric_exp);
- }
- // Weight quantile noise with modeled noise
- noise[i] *= (inst->blockInd);
- tmpFloat2 =
- inst->parametricNoise[i] * (END_STARTUP_SHORT - inst->blockInd);
- noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
- noise[i] /= END_STARTUP_SHORT;
- }
+ // Weight quantile noise with modeled noise
+ noise[i] *= (inst->blockInd);
+ tmpFloat2 =
+ inst->parametricNoise[i] * (END_STARTUP_SHORT - inst->blockInd);
+ noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
+ noise[i] /= END_STARTUP_SHORT;
}
- // compute average signal during END_STARTUP_LONG time:
- // used to normalize spectral difference measure
- if (inst->blockInd < END_STARTUP_LONG) {
- inst->featureData[5] *= inst->blockInd;
- inst->featureData[5] += signalEnergy;
- inst->featureData[5] /= (inst->blockInd + 1);
- }
+ }
+ // compute average signal during END_STARTUP_LONG time:
+ // used to normalize spectral difference measure
+ if (inst->blockInd < END_STARTUP_LONG) {
+ inst->featureData[5] *= inst->blockInd;
+ inst->featureData[5] += signalEnergy;
+ inst->featureData[5] /= (inst->blockInd + 1);
+ }
- // start processing at frames == converged+1
- // STEP 1: compute prior and post snr based on quantile noise est
- // compute DD estimate of prior SNR: needed for new method
- for (i = 0; i < inst->magnLen; i++) {
- // post snr
- snrLocPost[i] = (float)0.0;
- if (magn[i] > noise[i]) {
- snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
- }
- // previous post snr
- // previous estimate: based on previous frame with gain filter
- inst->previousEstimateStsa[i] = inst->magnPrev[i] /
- (inst->noisePrev[i] + (float)0.0001) *
- (inst->smooth[i]);
- // DD estimate is sum of two terms: current estimate and previous estimate
- // directed decision update of snrPrior
- snrLocPrior[i] = DD_PR_SNR * inst->previousEstimateStsa[i] +
- ((float)1.0 - DD_PR_SNR) * snrLocPost[i];
- // post and prior snr needed for step 2
- } // end of loop over freqs
- // done with step 1: dd computation of prior and post snr
+ // start processing at frames == converged+1
+ // STEP 1: compute prior and post snr based on quantile noise est
+ // compute DD estimate of prior SNR: needed for new method
+ for (i = 0; i < inst->magnLen; i++) {
+ // post snr
+ snrLocPost[i] = (float)0.0;
+ if (magn[i] > noise[i]) {
+ snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
+ }
+ // previous post snr
+ // previous estimate: based on previous frame with gain filter
+ inst->previousEstimateStsa[i] = inst->magnPrev[i] /
+ (inst->noisePrev[i] + (float)0.0001) *
+ (inst->smooth[i]);
+ // DD estimate is sum of two terms: current estimate and previous estimate
+ // directed decision update of snrPrior
+ snrLocPrior[i] = DD_PR_SNR * inst->previousEstimateStsa[i] +
+ ((float)1.0 - DD_PR_SNR) * snrLocPost[i];
+ // post and prior snr needed for step 2
+ } // end of loop over freqs
+ // done with step 1: dd computation of prior and post snr
- // STEP 2: compute speech/noise likelihood
- // compute difference of input spectrum with learned/estimated noise
- // spectrum
- WebRtcNs_ComputeSpectralDifference(inst, magn);
- // compute histograms for parameter decisions (thresholds and weights for
- // features)
- // parameters are extracted once every window time
- // (=inst->modelUpdatePars[1])
- if (updateParsFlag >= 1) {
- // counter update
- inst->modelUpdatePars[3]--;
- // update histogram
- if (inst->modelUpdatePars[3] > 0) {
- WebRtcNs_FeatureParameterExtraction(inst, 0);
- }
- // compute model parameters
- if (inst->modelUpdatePars[3] == 0) {
- WebRtcNs_FeatureParameterExtraction(inst, 1);
- inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
- // if wish to update only once, set flag to zero
- if (updateParsFlag == 1) {
- inst->modelUpdatePars[0] = 0;
- } else {
- // update every window:
- // get normalization for spectral difference for next window estimate
- inst->featureData[6] =
- inst->featureData[6] / ((float)inst->modelUpdatePars[1]);
- inst->featureData[5] =
- (float)0.5 * (inst->featureData[6] + inst->featureData[5]);
- inst->featureData[6] = (float)0.0;
- }
+ // STEP 2: compute speech/noise likelihood
+ // compute difference of input spectrum with learned/estimated noise
+ // spectrum
+ WebRtcNs_ComputeSpectralDifference(inst, magn);
+ // compute histograms for parameter decisions (thresholds and weights for
+ // features)
+ // parameters are extracted once every window time
+ // (=inst->modelUpdatePars[1])
+ if (updateParsFlag >= 1) {
+ // counter update
+ inst->modelUpdatePars[3]--;
+ // update histogram
+ if (inst->modelUpdatePars[3] > 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 0);
+ }
+ // compute model parameters
+ if (inst->modelUpdatePars[3] == 0) {
+ WebRtcNs_FeatureParameterExtraction(inst, 1);
+ inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
+ // if wish to update only once, set flag to zero
+ if (updateParsFlag == 1) {
+ inst->modelUpdatePars[0] = 0;
+ } else {
+ // update every window:
+ // get normalization for spectral difference for next window estimate
+ inst->featureData[6] =
+ inst->featureData[6] / ((float)inst->modelUpdatePars[1]);
+ inst->featureData[5] =
+ (float)0.5 * (inst->featureData[6] + inst->featureData[5]);
+ inst->featureData[6] = (float)0.0;
}
}
- // compute speech/noise probability
- WebRtcNs_SpeechNoiseProb(inst, inst->speechProb, snrLocPrior, snrLocPost);
- // time-avg parameter for noise update
+ }
+ // compute speech/noise probability
+ WebRtcNs_SpeechNoiseProb(inst, inst->speechProb, snrLocPrior, snrLocPost);
+ // time-avg parameter for noise update
+ gammaNoiseTmp = NOISE_UPDATE;
+ for (i = 0; i < inst->magnLen; i++) {
+ probSpeech = inst->speechProb[i];
+ probNonSpeech = (float)1.0 - probSpeech;
+ // temporary noise update:
+ // use it for speech frames if update value is less than previous
+ noiseUpdateTmp =
+ gammaNoiseTmp * inst->noisePrev[i] +
+ ((float)1.0 - gammaNoiseTmp) *
+ (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
+ //
+ // time-constant based on speech/noise state
+ gammaNoiseOld = gammaNoiseTmp;
gammaNoiseTmp = NOISE_UPDATE;
- for (i = 0; i < inst->magnLen; i++) {
- probSpeech = inst->speechProb[i];
- probNonSpeech = (float)1.0 - probSpeech;
- // temporary noise update:
- // use it for speech frames if update value is less than previous
- noiseUpdateTmp =
+ // increase gamma (i.e., less noise update) for frame likely to be speech
+ if (probSpeech > PROB_RANGE) {
+ gammaNoiseTmp = SPEECH_UPDATE;
+ }
+ // conservative noise update
+ if (probSpeech < PROB_RANGE) {
+ inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
+ }
+ // noise update
+ if (gammaNoiseTmp == gammaNoiseOld) {
+ noise[i] = noiseUpdateTmp;
+ } else {
+ noise[i] =
gammaNoiseTmp * inst->noisePrev[i] +
((float)1.0 - gammaNoiseTmp) *
(probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- //
- // time-constant based on speech/noise state
- gammaNoiseOld = gammaNoiseTmp;
- gammaNoiseTmp = NOISE_UPDATE;
- // increase gamma (i.e., less noise update) for frame likely to be speech
- if (probSpeech > PROB_RANGE) {
- gammaNoiseTmp = SPEECH_UPDATE;
- }
- // conservative noise update
- if (probSpeech < PROB_RANGE) {
- inst->magnAvgPause[i] +=
- GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
- }
- // noise update
- if (gammaNoiseTmp == gammaNoiseOld) {
+ // allow for noise update downwards:
+ // if noise update decreases the noise, it is safe, so allow it to
+ // happen
+ if (noiseUpdateTmp < noise[i]) {
noise[i] = noiseUpdateTmp;
- } else {
- noise[i] =
- gammaNoiseTmp * inst->noisePrev[i] +
- ((float)1.0 - gammaNoiseTmp) *
- (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- // allow for noise update downwards:
- // if noise update decreases the noise, it is safe, so allow it to
- // happen
- if (noiseUpdateTmp < noise[i]) {
- noise[i] = noiseUpdateTmp;
- }
}
- } // end of freq loop
- // done with step 2: noise update
-
- // keep track of noise spectrum for next frame
- for (i = 0; i < inst->magnLen; i++) {
- inst->noisePrev[i] = noise[i];
}
- } // end of if inst->outLen == 0
+ } // end of freq loop
+ // done with step 2: noise update
+
+ // keep track of noise spectrum for next frame
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->noisePrev[i] = noise[i];
+ }
return 0;
}
@@ -1081,194 +1068,30 @@
// update analysis buffer for L band
memcpy(inst->dataBuf,
- inst->dataBuf + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms,
+ inst->dataBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memcpy(inst->dataBuf + inst->anaLen - inst->blockLen,
speechFrame,
- sizeof(float) * inst->blockLen10ms);
+ sizeof(float) * inst->blockLen);
if (flagHB == 1) {
// update analysis buffer for H band
memcpy(inst->dataBufHB,
- inst->dataBufHB + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms,
+ inst->dataBufHB + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen,
speechFrameHB,
- sizeof(float) * inst->blockLen10ms);
+ sizeof(float) * inst->blockLen);
}
- // check if processing needed
- if (inst->outLen == 0) {
- // windowing
- energy1 = 0.0;
- for (i = 0; i < inst->anaLen; i++) {
- winData[i] = inst->window[i] * inst->dataBuf[i];
- energy1 += winData[i] * winData[i];
- }
- if (energy1 == 0.0) {
- // synthesize the special case of zero input
- // read out fully processed segment
- for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
- fout[i - inst->windShift] = inst->syntBuf[i];
- }
- // update synthesis buffer
- memcpy(inst->syntBuf,
- inst->syntBuf + inst->blockLen,
- sizeof(float) * (inst->anaLen - inst->blockLen));
- memset(inst->syntBuf + inst->anaLen - inst->blockLen,
- 0,
- sizeof(float) * inst->blockLen);
-
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms) {
- for (i = 0; i < inst->outLen; i++) {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
- }
- for (i = 0; i < inst->blockLen10ms; ++i)
- outFrame[i] = WEBRTC_SPL_SAT(
- WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
-
- // for time-domain gain of HB
- if (flagHB == 1)
- for (i = 0; i < inst->blockLen10ms; ++i)
- outFrameHB[i] = WEBRTC_SPL_SAT(
- WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN);
-
- return 0;
- }
-
- // FFT
- WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
-
- imag[0] = 0;
- real[0] = winData[0];
- magn[0] = (float)(fabs(real[0]) + 1.0f);
- imag[inst->magnLen - 1] = 0;
- real[inst->magnLen - 1] = winData[1];
- magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
- if (inst->blockInd < END_STARTUP_SHORT) {
- inst->initMagnEst[0] += magn[0];
- inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
- }
- for (i = 1; i < inst->magnLen - 1; i++) {
- real[i] = winData[2 * i];
- imag[i] = winData[2 * i + 1];
- // magnitude spectrum
- fTmp = real[i] * real[i];
- fTmp += imag[i] * imag[i];
- magn[i] = ((float)sqrt(fTmp)) + 1.0f;
- if (inst->blockInd < END_STARTUP_SHORT) {
- inst->initMagnEst[i] += magn[i];
- }
- }
-
- // Compute dd update of prior snr and post snr based on new noise estimate
- for (i = 0; i < inst->magnLen; i++) {
- // post and prior snr
- currentEstimateStsa = (float)0.0;
- if (magn[i] > inst->noisePrev[i]) {
- currentEstimateStsa =
- magn[i] / (inst->noisePrev[i] + (float)0.0001) - (float)1.0;
- }
- // DD estimate is sume of two terms: current estimate and previous
- // estimate
- // directed decision update of snrPrior
- snrPrior = DD_PR_SNR * inst->previousEstimateStsa[i] +
- ((float)1.0 - DD_PR_SNR) * currentEstimateStsa;
- // gain filter
- tmpFloat1 = inst->overdrive + snrPrior;
- tmpFloat2 = (float)snrPrior / tmpFloat1;
- theFilter[i] = (float)tmpFloat2;
- } // end of loop over freqs
-
- for (i = 0; i < inst->magnLen; i++) {
- // flooring bottom
- if (theFilter[i] < inst->denoiseBound) {
- theFilter[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilter[i] > (float)1.0) {
- theFilter[i] = 1.0;
- }
- if (inst->blockInd < END_STARTUP_SHORT) {
- theFilterTmp[i] =
- (inst->initMagnEst[i] - inst->overdrive * inst->parametricNoise[i]);
- theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
- // flooring bottom
- if (theFilterTmp[i] < inst->denoiseBound) {
- theFilterTmp[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilterTmp[i] > (float)1.0) {
- theFilterTmp[i] = 1.0;
- }
- // Weight the two suppression filters
- theFilter[i] *= (inst->blockInd);
- theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
- theFilter[i] += theFilterTmp[i];
- theFilter[i] /= (END_STARTUP_SHORT);
- }
- // smoothing
- inst->smooth[i] = theFilter[i];
- real[i] *= inst->smooth[i];
- imag[i] *= inst->smooth[i];
- }
- // keep track of magn spectrum for next frame
- for (i = 0; i < inst->magnLen; i++) {
- inst->magnPrev[i] = magn[i];
- }
- // back to time domain
- winData[0] = real[0];
- winData[1] = real[inst->magnLen - 1];
- for (i = 1; i < inst->magnLen - 1; i++) {
- winData[2 * i] = real[i];
- winData[2 * i + 1] = imag[i];
- }
- WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
-
- for (i = 0; i < inst->anaLen; i++) {
- real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
- }
-
- // scale factor: only do it after END_STARTUP_LONG time
- factor = (float)1.0;
- if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) {
- factor1 = (float)1.0;
- factor2 = (float)1.0;
-
- energy2 = 0.0;
- for (i = 0; i < inst->anaLen; i++) {
- energy2 += (float)real[i] * (float)real[i];
- }
- gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
-
- // scaling for new version
- if (gain > B_LIM) {
- factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
- if (gain * factor1 > (float)1.0) {
- factor1 = (float)1.0 / gain;
- }
- }
- if (gain < B_LIM) {
- // don't reduce scale too much for pause regions:
- // attenuation here should be controlled by flooring
- if (gain <= inst->denoiseBound) {
- gain = inst->denoiseBound;
- }
- factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
- }
- // combine both scales with speech/noise prob:
- // note prior (priorSpeechProb) is not frequency dependent
- factor = inst->priorSpeechProb * factor1 +
- ((float)1.0 - inst->priorSpeechProb) * factor2;
- } // out of inst->gainmap==1
-
- // synthesis
- for (i = 0; i < inst->anaLen; i++) {
- inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
- }
+ // windowing
+ energy1 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ winData[i] = inst->window[i] * inst->dataBuf[i];
+ energy1 += winData[i] * winData[i];
+ }
+ if (energy1 == 0.0) {
+ // synthesize the special case of zero input
// read out fully processed segment
for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
fout[i - inst->windShift] = inst->syntBuf[i];
@@ -1281,28 +1104,161 @@
0,
sizeof(float) * inst->blockLen);
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms) {
- for (i = 0; i < inst->outLen; i++) {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
+ for (i = 0; i < inst->blockLen; ++i)
+ outFrame[i] =
+ WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
+
+ // for time-domain gain of HB
+ if (flagHB == 1)
+ for (i = 0; i < inst->blockLen; ++i)
+ outFrameHB[i] = WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN);
+
+ return 0;
+ }
+ // FFT
+ WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
+
+ imag[0] = 0;
+ real[0] = winData[0];
+ magn[0] = (float)(fabs(real[0]) + 1.0f);
+ imag[inst->magnLen - 1] = 0;
+ real[inst->magnLen - 1] = winData[1];
+ magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[0] += magn[0];
+ inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
+ }
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ real[i] = winData[2 * i];
+ imag[i] = winData[2 * i + 1];
+ // magnitude spectrum
+ fTmp = real[i] * real[i];
+ fTmp += imag[i] * imag[i];
+ magn[i] = ((float)sqrt(fTmp)) + 1.0f;
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ inst->initMagnEst[i] += magn[i];
}
- } // end of if out.len==0
- else {
- for (i = 0; i < inst->blockLen10ms; i++) {
- fout[i] = inst->outBuf[i];
- }
- memcpy(inst->outBuf,
- inst->outBuf + inst->blockLen10ms,
- sizeof(float) * (inst->outLen - inst->blockLen10ms));
- memset(inst->outBuf + inst->outLen - inst->blockLen10ms,
- 0,
- sizeof(float) * inst->blockLen10ms);
- inst->outLen -= inst->blockLen10ms;
}
- for (i = 0; i < inst->blockLen10ms; ++i)
+ // Compute dd update of prior snr and post snr based on new noise estimate
+ for (i = 0; i < inst->magnLen; i++) {
+ // post and prior snr
+ currentEstimateStsa = (float)0.0;
+ if (magn[i] > inst->noisePrev[i]) {
+ currentEstimateStsa =
+ magn[i] / (inst->noisePrev[i] + (float)0.0001) - (float)1.0;
+ }
+ // DD estimate is sume of two terms: current estimate and previous
+ // estimate
+ // directed decision update of snrPrior
+ snrPrior = DD_PR_SNR * inst->previousEstimateStsa[i] +
+ ((float)1.0 - DD_PR_SNR) * currentEstimateStsa;
+ // gain filter
+ tmpFloat1 = inst->overdrive + snrPrior;
+ tmpFloat2 = (float)snrPrior / tmpFloat1;
+ theFilter[i] = (float)tmpFloat2;
+ } // end of loop over freqs
+
+ for (i = 0; i < inst->magnLen; i++) {
+ // flooring bottom
+ if (theFilter[i] < inst->denoiseBound) {
+ theFilter[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilter[i] > (float)1.0) {
+ theFilter[i] = 1.0;
+ }
+ if (inst->blockInd < END_STARTUP_SHORT) {
+ theFilterTmp[i] =
+ (inst->initMagnEst[i] - inst->overdrive * inst->parametricNoise[i]);
+ theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
+ // flooring bottom
+ if (theFilterTmp[i] < inst->denoiseBound) {
+ theFilterTmp[i] = inst->denoiseBound;
+ }
+ // flooring top
+ if (theFilterTmp[i] > (float)1.0) {
+ theFilterTmp[i] = 1.0;
+ }
+ // Weight the two suppression filters
+ theFilter[i] *= (inst->blockInd);
+ theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
+ theFilter[i] += theFilterTmp[i];
+ theFilter[i] /= (END_STARTUP_SHORT);
+ }
+ // smoothing
+ inst->smooth[i] = theFilter[i];
+ real[i] *= inst->smooth[i];
+ imag[i] *= inst->smooth[i];
+ }
+ // keep track of magn spectrum for next frame
+ for (i = 0; i < inst->magnLen; i++) {
+ inst->magnPrev[i] = magn[i];
+ }
+ // back to time domain
+ winData[0] = real[0];
+ winData[1] = real[inst->magnLen - 1];
+ for (i = 1; i < inst->magnLen - 1; i++) {
+ winData[2 * i] = real[i];
+ winData[2 * i + 1] = imag[i];
+ }
+ WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
+
+ for (i = 0; i < inst->anaLen; i++) {
+ real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
+ }
+
+ // scale factor: only do it after END_STARTUP_LONG time
+ factor = (float)1.0;
+ if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) {
+ factor1 = (float)1.0;
+ factor2 = (float)1.0;
+
+ energy2 = 0.0;
+ for (i = 0; i < inst->anaLen; i++) {
+ energy2 += (float)real[i] * (float)real[i];
+ }
+ gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
+
+ // scaling for new version
+ if (gain > B_LIM) {
+ factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
+ if (gain * factor1 > (float)1.0) {
+ factor1 = (float)1.0 / gain;
+ }
+ }
+ if (gain < B_LIM) {
+ // don't reduce scale too much for pause regions:
+ // attenuation here should be controlled by flooring
+ if (gain <= inst->denoiseBound) {
+ gain = inst->denoiseBound;
+ }
+ factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
+ }
+ // combine both scales with speech/noise prob:
+ // note prior (priorSpeechProb) is not frequency dependent
+ factor = inst->priorSpeechProb * factor1 +
+ ((float)1.0 - inst->priorSpeechProb) * factor2;
+ } // out of inst->gainmap==1
+
+ // synthesis
+ for (i = 0; i < inst->anaLen; i++) {
+ inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
+ }
+ // read out fully processed segment
+ for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
+ fout[i - inst->windShift] = inst->syntBuf[i];
+ }
+ // update synthesis buffer
+ memcpy(inst->syntBuf,
+ inst->syntBuf + inst->blockLen,
+ sizeof(float) * (inst->anaLen - inst->blockLen));
+ memset(inst->syntBuf + inst->anaLen - inst->blockLen,
+ 0,
+ sizeof(float) * inst->blockLen);
+
+ for (i = 0; i < inst->blockLen; ++i)
outFrame[i] =
WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
@@ -1343,7 +1299,7 @@
gainTimeDomainHB = 1.0;
}
// apply gain
- for (i = 0; i < inst->blockLen10ms; i++) {
+ for (i = 0; i < inst->blockLen; i++) {
float o = gainTimeDomainHB * inst->dataBufHB[i];
outFrameHB[i] =
WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN);
diff --git a/webrtc/modules/audio_processing/ns/ns_core.h b/webrtc/modules/audio_processing/ns/ns_core.h
index c5ca13f..2d36d8a 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.h
+++ b/webrtc/modules/audio_processing/ns/ns_core.h
@@ -52,9 +52,7 @@
typedef struct NSinst_t_ {
uint32_t fs;
int blockLen;
- int blockLen10ms;
int windShift;
- int outLen;
int anaLen;
int magnLen;
int aggrMode;
@@ -62,7 +60,6 @@
float analyzeBuf[ANAL_BLOCKL_MAX];
float dataBuf[ANAL_BLOCKL_MAX];
float syntBuf[ANAL_BLOCKL_MAX];
- float outBuf[3 * BLOCKL_MAX];
int initFlag;
// parameters for quantile noise estimation
