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Noise suppression: Change signature to work on floats instead of ints
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Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6385 4adac7df-926f-26a2-2b94-8c16560cd09d
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kwiberg@webrtc.org committed Jun 10, 2014
1 parent 1014101 commit 12cd443
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Showing 6 changed files with 36 additions and 70 deletions.
Binary file modified data/audio_processing/output_data_float.pb
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8 changes: 4 additions & 4 deletions webrtc/modules/audio_processing/noise_suppression_impl.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -68,10 +68,10 @@ int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
Handle* my_handle = static_cast<Handle*>(handle(i));
#if defined(WEBRTC_NS_FLOAT)
err = WebRtcNs_Process(static_cast<Handle*>(handle(i)),
audio->low_pass_split_data(i),
audio->high_pass_split_data(i),
audio->low_pass_split_data(i),
audio->high_pass_split_data(i));
audio->low_pass_split_data_f(i),
audio->high_pass_split_data_f(i),
audio->low_pass_split_data_f(i),
audio->high_pass_split_data_f(i));
#elif defined(WEBRTC_NS_FIXED)
err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)),
audio->low_pass_split_data(i),
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8 changes: 4 additions & 4 deletions webrtc/modules/audio_processing/ns/include/noise_suppression.h
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Original file line number Diff line number Diff line change
Expand Up @@ -99,10 +99,10 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode);
* -1 - Error
*/
int WebRtcNs_Process(NsHandle* NS_inst,
short* spframe,
short* spframe_H,
short* outframe,
short* outframe_H);
float* spframe,
float* spframe_H,
float* outframe,
float* outframe_H);

/* Returns the internally used prior speech probability of the current frame.
* There is a frequency bin based one as well, with which this should not be
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4 changes: 2 additions & 2 deletions webrtc/modules/audio_processing/ns/noise_suppression.c
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Original file line number Diff line number Diff line change
Expand Up @@ -43,8 +43,8 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
}


int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H,
short* outframe, short* outframe_H) {
int WebRtcNs_Process(NsHandle* NS_inst, float* spframe, float* spframe_H,
float* outframe, float* outframe_H) {
return WebRtcNs_ProcessCore(
(NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
}
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78 changes: 22 additions & 56 deletions webrtc/modules/audio_processing/ns/ns_core.c
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Original file line number Diff line number Diff line change
Expand Up @@ -715,10 +715,10 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr
}

int WebRtcNs_ProcessCore(NSinst_t* inst,
short* speechFrame,
short* speechFrameHB,
short* outFrame,
short* outFrameHB) {
float* speechFrame,
float* speechFrameHB,
float* outFrame,
float* outFrameHB) {
// main routine for noise reduction

int flagHB = 0;
Expand All @@ -731,8 +731,8 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
float snrPrior, currentEstimateStsa;
float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
float gammaNoiseTmp, gammaNoiseOld;
float noiseUpdateTmp, fTmp, dTmp;
float fin[BLOCKL_MAX], fout[BLOCKL_MAX];
float noiseUpdateTmp, fTmp;
float fout[BLOCKL_MAX];
float winData[ANAL_BLOCKL_MAX];
float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
Expand Down Expand Up @@ -775,26 +775,17 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
updateParsFlag = inst->modelUpdatePars[0];
//

//for LB do all processing
// convert to float
for (i = 0; i < inst->blockLen10ms; i++) {
fin[i] = (float)speechFrame[i];
}
// 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, fin,
memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, speechFrame,
sizeof(float) * inst->blockLen10ms);

if (flagHB == 1) {
// convert to float
for (i = 0; i < inst->blockLen10ms; i++) {
fin[i] = (float)speechFrameHB[i];
}
// 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, fin,
memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, speechFrameHB,
sizeof(float) * inst->blockLen10ms);
}

Expand Down Expand Up @@ -833,30 +824,16 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
inst->outBuf[i] = fout[i + inst->blockLen10ms];
}
}
// convert to short
for (i = 0; i < inst->blockLen10ms; i++) {
dTmp = fout[i];
if (dTmp < WEBRTC_SPL_WORD16_MIN) {
dTmp = WEBRTC_SPL_WORD16_MIN;
} else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
dTmp = WEBRTC_SPL_WORD16_MAX;
}
outFrame[i] = (short)dTmp;
}
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++) {
dTmp = inst->dataBufHB[i];
if (dTmp < WEBRTC_SPL_WORD16_MIN) {
dTmp = WEBRTC_SPL_WORD16_MIN;
} else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
dTmp = WEBRTC_SPL_WORD16_MAX;
}
outFrameHB[i] = (short)dTmp;
}
} // end of H band gain computation
//
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;
}

Expand Down Expand Up @@ -1239,16 +1216,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
inst->outLen -= inst->blockLen10ms;
}

// convert to short
for (i = 0; i < inst->blockLen10ms; i++) {
dTmp = fout[i];
if (dTmp < WEBRTC_SPL_WORD16_MIN) {
dTmp = WEBRTC_SPL_WORD16_MIN;
} else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
dTmp = WEBRTC_SPL_WORD16_MAX;
}
outFrame[i] = (short)dTmp;
}
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) {
Expand Down Expand Up @@ -1289,13 +1259,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
}
//apply gain
for (i = 0; i < inst->blockLen10ms; i++) {
dTmp = gainTimeDomainHB * inst->dataBufHB[i];
if (dTmp < WEBRTC_SPL_WORD16_MIN) {
dTmp = WEBRTC_SPL_WORD16_MIN;
} else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
dTmp = WEBRTC_SPL_WORD16_MAX;
}
outFrameHB[i] = (short)dTmp;
float o = gainTimeDomainHB * inst->dataBufHB[i];
outFrameHB[i] = WEBRTC_SPL_SAT(
WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN);
}
} // end of H band gain computation
//
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8 changes: 4 additions & 4 deletions webrtc/modules/audio_processing/ns/ns_core.h
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Original file line number Diff line number Diff line change
Expand Up @@ -167,10 +167,10 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode);


int WebRtcNs_ProcessCore(NSinst_t* inst,
short* inFrameLow,
short* inFrameHigh,
short* outFrameLow,
short* outFrameHigh);
float* inFrameLow,
float* inFrameHigh,
float* outFrameLow,
float* outFrameHigh);


#ifdef __cplusplus
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