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Remove the different block lengths in ns_core

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This CL has bit-exact output.

What it does:
  * Remove the blockLen10Ms, as it is hardcoded to be equal to blockLen.
  * This makes outLen to be always zero, so it can be removed too.
  * It also avoids the need to have an outBuf, because it is not used, so it is also removed
  * Replaced blockLen10Ms by blockLen everywhere, since they were hardcoded to be equal.
  * We don't need to check if outLen is zero, because it always is, so it was removed.
  * Of course, the outBuf needs no initial set or copying around, because it is not used.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7297 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2014-09-25 13:53:43 +00:00
parent 741711a861
commit 5f3965783b
5 changed files with 402 additions and 439 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
webrtc/modules/audio_processing/noise_suppression_impl.cc
View File

@ -173,7 +173,8 @@ void NoiseSuppressionImpl::DestroyHandle(void* handle) const {
int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
#if defined(WEBRTC_NS_FLOAT)
return WebRtcNs_Init(static_cast<Handle*>(handle),
apm_->proc_sample_rate_hz());
apm_->proc_sample_rate_hz(),
AudioProcessing::kChunkSizeMs);
#elif defined(WEBRTC_NS_FIXED)
return WebRtcNsx_Init(static_cast<Handle*>(handle),
apm_->proc_sample_rate_hz());

3
webrtc/modules/audio_processing/ns/include/noise_suppression.h
View File

@ -55,6 +55,7 @@ int WebRtcNs_Free(NsHandle* NS_inst);
* Input:
* - NS_inst : Instance that should be initialized
* - fs : sampling frequency
* - blockLenMs : block length in ms
*
* Output:
* - NS_inst : Initialized instance
@ -62,7 +63,7 @@ int WebRtcNs_Free(NsHandle* NS_inst);
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcNs_Init(NsHandle* NS_inst, uint32_t fs);
int WebRtcNs_Init(NsHandle* NS_inst, uint32_t fs, int blockLenMs);
/*
* This changes the aggressiveness of the noise suppression method.

4
webrtc/modules/audio_processing/ns/noise_suppression.c
View File

@ -34,8 +34,8 @@ int WebRtcNs_Free(NsHandle* NS_inst) {
}
int WebRtcNs_Init(NsHandle* NS_inst, uint32_t fs) {
return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
int WebRtcNs_Init(NsHandle* NS_inst, uint32_t fs, int blockLenMs) {
return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs, blockLenMs);
}
int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {

80
webrtc/modules/audio_processing/ns/ns_core.c
View File

@ -71,7 +71,7 @@ void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) {
}
// Initialize state
int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) {
int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs, int blockLenMs) {
int i;
// We only support 10ms frames
@ -89,27 +89,22 @@ int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs) {
inst->windShift = 0;
if (fs == 8000) {
// We only support 10ms frames
inst->blockLen = 80;
inst->blockLen10ms = 80;
inst->blockLen = 8;
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->blockLen = 16;
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->blockLen = 16;
inst->anaLen = 256;
inst->window = kBlocks160w256;
inst->outLen = 0;
}
inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
inst->blockLen *= blockLenMs;
// Initialize fft work arrays.
inst->ip[0] = 0; // Setting this triggers initialization.
@ -789,14 +784,12 @@ int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* speechFrame) {
// 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++) {
@ -1032,7 +1025,6 @@ int WebRtcNs_AnalyzeCore(NSinst_t* inst, float* speechFrame) {
for (i = 0; i < inst->magnLen; i++) {
inst->noisePrev[i] = noise[i];
}
} // end of if inst->outLen == 0
return 0;
}
@ -1081,24 +1073,22 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
// 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++) {
@ -1119,20 +1109,13 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
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)
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->blockLen10ms; ++i)
for (i = 0; i < inst->blockLen; ++i)
outFrameHB[i] = WEBRTC_SPL_SAT(
WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN);
@ -1281,28 +1264,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
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];
}
}
} // 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)
for (i = 0; i < inst->blockLen; ++i)
outFrame[i] =
WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
@ -1343,7 +1305,7 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
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);

5
webrtc/modules/audio_processing/ns/ns_core.h
View File

@ -52,9 +52,7 @@ typedef struct NSParaExtract_t_ {
typedef struct NSinst_t_ {
uint32_t fs;
int blockLen;
int blockLen10ms;
int windShift;
int outLen;
int anaLen;
int magnLen;
int aggrMode;
@ -122,6 +120,7 @@ extern "C" {
* Input:
* - inst : Instance that should be initialized
* - fs : Sampling frequency
* - blockLenMs : Block length in ms
*
* Output:
* - inst : Initialized instance
@ -129,7 +128,7 @@ extern "C" {
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs);
int WebRtcNs_InitCore(NSinst_t* inst, uint32_t fs, int blockLenMs);
/****************************************************************************
* WebRtcNs_set_policy_core(...)