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Revert "Revert "Audio processing: Feed each processing step its choice
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of int or float data"

This reverts commit 6142.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
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mflodman@webrtc.org committed May 15, 2014
1 parent 024e4d5 commit d5da250
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Showing 9 changed files with 180 additions and 95 deletions.
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59 changes: 23 additions & 36 deletions webrtc/modules/audio_processing/aec/aec_core.c
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Original file line number Diff line number Diff line change
Expand Up @@ -116,7 +116,7 @@ extern int webrtc_aec_instance_count;
// "Private" function prototypes.
static void ProcessBlock(AecCore* aec);

static void NonLinearProcessing(AecCore* aec, short* output, short* outputH);
static void NonLinearProcessing(AecCore* aec, float* output, float* outputH);

static void GetHighbandGain(const float* lambda, float* nlpGainHband);

Expand Down Expand Up @@ -160,28 +160,28 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
return -1;
}

aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
if (!aec->nearFrBuf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}

aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
if (!aec->outFrBuf) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}

aec->nearFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
aec->nearFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
if (!aec->nearFrBufH) {
WebRtcAec_FreeAec(aec);
aec = NULL;
return -1;
}

aec->outFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
aec->outFrBufH = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
if (!aec->outFrBufH) {
WebRtcAec_FreeAec(aec);
aec = NULL;
Expand Down Expand Up @@ -617,11 +617,11 @@ int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) {
}

void WebRtcAec_ProcessFrame(AecCore* aec,
const short* nearend,
const short* nearendH,
const float* nearend,
const float* nearendH,
int knownDelay,
int16_t* out,
int16_t* outH) {
float* out,
float* outH) {
int out_elements = 0;

// For each frame the process is as follows:
Expand Down Expand Up @@ -814,7 +814,7 @@ void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {

static void ProcessBlock(AecCore* aec) {
int i;
float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
float y[PART_LEN], e[PART_LEN];
float scale;

float fft[PART_LEN2];
Expand All @@ -833,30 +833,22 @@ static void ProcessBlock(AecCore* aec) {
const float ramp = 1.0002f;
const float gInitNoise[2] = {0.999f, 0.001f};

int16_t nearend[PART_LEN];
int16_t* nearend_ptr = NULL;
int16_t output[PART_LEN];
int16_t outputH[PART_LEN];
float nearend[PART_LEN];
float* nearend_ptr = NULL;
float output[PART_LEN];
float outputH[PART_LEN];

float* xf_ptr = NULL;

memset(dH, 0, sizeof(dH));
// Concatenate old and new nearend blocks.
if (aec->sampFreq == 32000) {
// Get the upper band first so we can reuse |nearend|.
WebRtc_ReadBuffer(aec->nearFrBufH, (void**)&nearend_ptr, nearend, PART_LEN);
for (i = 0; i < PART_LEN; i++) {
dH[i] = (float)(nearend_ptr[i]);
}
memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
memcpy(aec->dBufH + PART_LEN, nearend_ptr, sizeof(nearend));
}
WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN);
memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend));

// ---------- Ooura fft ----------
// Concatenate old and new nearend blocks.
for (i = 0; i < PART_LEN; i++) {
d[i] = (float)(nearend_ptr[i]);
}
memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);

#ifdef WEBRTC_AEC_DEBUG_DUMP
{
Expand Down Expand Up @@ -968,7 +960,7 @@ static void ProcessBlock(AecCore* aec) {
}

for (i = 0; i < PART_LEN; i++) {
e[i] = d[i] - y[i];
e[i] = nearend_ptr[i] - y[i];
}

// Error fft
Expand Down Expand Up @@ -1027,7 +1019,7 @@ static void ProcessBlock(AecCore* aec) {
#endif
}

static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) {
static void NonLinearProcessing(AecCore* aec, float* output, float* outputH) {
float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1];
complex_t comfortNoiseHband[PART_LEN1];
float fft[PART_LEN2];
Expand Down Expand Up @@ -1321,13 +1313,10 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) {
fft[i] *= scale; // fft scaling
fft[i] = fft[i] * sqrtHanning[i] + aec->outBuf[i];

// Saturation protection
output[i] = (short)WEBRTC_SPL_SAT(
WEBRTC_SPL_WORD16_MAX, fft[i], WEBRTC_SPL_WORD16_MIN);

fft[PART_LEN + i] *= scale; // fft scaling
aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i];
}
memcpy(output, fft, sizeof(*output) * PART_LEN);

// For H band
if (aec->sampFreq == 32000) {
Expand All @@ -1351,18 +1340,16 @@ static void NonLinearProcessing(AecCore* aec, short* output, short* outputH) {

// compute gain factor
for (i = 0; i < PART_LEN; i++) {
dtmp = (float)aec->dBufH[i];
dtmp = (float)dtmp * nlpGainHband; // for variable gain
dtmp = aec->dBufH[i];
dtmp = dtmp * nlpGainHband; // for variable gain

// add some comfort noise where Hband is attenuated
if (flagHbandCn == 1) {
fft[i] *= scale; // fft scaling
dtmp += cnScaleHband * fft[i];
}

// Saturation protection
outputH[i] = (short)WEBRTC_SPL_SAT(
WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
outputH[i] = dtmp;
}
}

Expand Down
8 changes: 4 additions & 4 deletions webrtc/modules/audio_processing/aec/aec_core.h
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Original file line number Diff line number Diff line change
Expand Up @@ -60,11 +60,11 @@ void WebRtcAec_InitAec_mips(void);

void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend);
void WebRtcAec_ProcessFrame(AecCore* aec,
const short* nearend,
const short* nearendH,
const float* nearend,
const float* nearendH,
int knownDelay,
int16_t* out,
int16_t* outH);
float* out,
float* outH);

// A helper function to call WebRtc_MoveReadPtr() for all far-end buffers.
// Returns the number of elements moved, and adjusts |system_delay| by the
Expand Down
48 changes: 24 additions & 24 deletions webrtc/modules/audio_processing/aec/echo_cancellation.c
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Original file line number Diff line number Diff line change
Expand Up @@ -104,18 +104,18 @@ int webrtc_aec_instance_count = 0;
static void EstBufDelayNormal(aecpc_t* aecInst);
static void EstBufDelayExtended(aecpc_t* aecInst);
static int ProcessNormal(aecpc_t* self,
const int16_t* near,
const int16_t* near_high,
int16_t* out,
int16_t* out_high,
const float* near,
const float* near_high,
float* out,
float* out_high,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
static void ProcessExtended(aecpc_t* self,
const int16_t* near,
const int16_t* near_high,
int16_t* out,
int16_t* out_high,
const float* near,
const float* near_high,
float* out,
float* out_high,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
Expand Down Expand Up @@ -372,10 +372,10 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
}

int32_t WebRtcAec_Process(void* aecInst,
const int16_t* nearend,
const int16_t* nearendH,
int16_t* out,
int16_t* outH,
const float* nearend,
const float* nearendH,
float* out,
float* outH,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew) {
Expand Down Expand Up @@ -632,10 +632,10 @@ AecCore* WebRtcAec_aec_core(void* handle) {
}

static int ProcessNormal(aecpc_t* aecpc,
const int16_t* nearend,
const int16_t* nearendH,
int16_t* out,
int16_t* outH,
const float* nearend,
const float* nearendH,
float* out,
float* outH,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew) {
Expand Down Expand Up @@ -689,10 +689,10 @@ static int ProcessNormal(aecpc_t* aecpc,
if (aecpc->startup_phase) {
// Only needed if they don't already point to the same place.
if (nearend != out) {
memcpy(out, nearend, sizeof(short) * nrOfSamples);
memcpy(out, nearend, sizeof(*out) * nrOfSamples);
}
if (nearendH != outH) {
memcpy(outH, nearendH, sizeof(short) * nrOfSamples);
memcpy(outH, nearendH, sizeof(*outH) * nrOfSamples);
}

// The AEC is in the start up mode
Expand Down Expand Up @@ -789,10 +789,10 @@ static int ProcessNormal(aecpc_t* aecpc,
}

static void ProcessExtended(aecpc_t* self,
const int16_t* near,
const int16_t* near_high,
int16_t* out,
int16_t* out_high,
const float* near,
const float* near_high,
float* out,
float* out_high,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew) {
Expand Down Expand Up @@ -823,10 +823,10 @@ static void ProcessExtended(aecpc_t* self,
if (!self->farend_started) {
// Only needed if they don't already point to the same place.
if (near != out) {
memcpy(out, near, sizeof(short) * num_samples);
memcpy(out, near, sizeof(*out) * num_samples);
}
if (near_high != out_high) {
memcpy(out_high, near_high, sizeof(short) * num_samples);
memcpy(out_high, near_high, sizeof(*out_high) * num_samples);
}
return;
}
Expand Down
16 changes: 8 additions & 8 deletions webrtc/modules/audio_processing/aec/include/echo_cancellation.h
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Original file line number Diff line number Diff line change
Expand Up @@ -133,9 +133,9 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
* Inputs Description
* -------------------------------------------------------------------
* void* aecInst Pointer to the AEC instance
* int16_t* nearend In buffer containing one frame of
* float* nearend In buffer containing one frame of
* nearend+echo signal for L band
* int16_t* nearendH In buffer containing one frame of
* float* nearendH In buffer containing one frame of
* nearend+echo signal for H band
* int16_t nrOfSamples Number of samples in nearend buffer
* int16_t msInSndCardBuf Delay estimate for sound card and
Expand All @@ -146,18 +146,18 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
*
* Outputs Description
* -------------------------------------------------------------------
* int16_t* out Out buffer, one frame of processed nearend
* float* out Out buffer, one frame of processed nearend
* for L band
* int16_t* outH Out buffer, one frame of processed nearend
* float* outH Out buffer, one frame of processed nearend
* for H band
* int32_t return 0: OK
* -1: error
*/
int32_t WebRtcAec_Process(void* aecInst,
const int16_t* nearend,
const int16_t* nearendH,
int16_t* out,
int16_t* outH,
const float* nearend,
const float* nearendH,
float* out,
float* outH,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew);
Expand Down
10 changes: 6 additions & 4 deletions webrtc/modules/audio_processing/aec/system_delay_unittest.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -46,16 +46,18 @@ class SystemDelayTest : public ::testing::Test {
aecpc_t* self_;
int samples_per_frame_;
// Dummy input/output speech data.
int16_t far_[160];
int16_t near_[160];
int16_t out_[160];
static const int kSamplesPerChunk = 160;
int16_t far_[kSamplesPerChunk];
float near_[kSamplesPerChunk];
float out_[kSamplesPerChunk];
};

SystemDelayTest::SystemDelayTest()
: handle_(NULL), self_(NULL), samples_per_frame_(0) {
// Dummy input data are set with more or less arbitrary non-zero values.
memset(far_, 1, sizeof(far_));
memset(near_, 2, sizeof(near_));
for (int i = 0; i < kSamplesPerChunk; i++)
near_[i] = 514.0;
memset(out_, 0, sizeof(out_));
}

Expand Down
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