Re-land "Support 48kHz in AEC"
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/
BUG=webrtc:3146
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c
index ef04997..e67273e 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -502,7 +502,7 @@
noiseAvg = 0.0;
tmpAvg = 0.0;
num = 0;
- if (aec->sampFreq == 32000 && flagHbandCn == 1) {
+ if (aec->num_bands > 1 && flagHbandCn == 1) {
// average noise scale
// average over second half of freq spectrum (i.e., 4->8khz)
@@ -849,13 +849,15 @@
return delay_correction;
}
-static void NonLinearProcessing(AecCore* aec, float* output, float* outputH) {
+static void NonLinearProcessing(AecCore* aec,
+ float* output,
+ float* const* outputH) {
float efw[2][PART_LEN1], xfw[2][PART_LEN1];
complex_t comfortNoiseHband[PART_LEN1];
float fft[PART_LEN2];
float scale, dtmp;
float nlpGainHband;
- int i;
+ int i, j;
// Coherence and non-linear filter
float cohde[PART_LEN1], cohxd[PART_LEN1];
@@ -1027,7 +1029,7 @@
}
// For H band
- if (aec->sampFreq == 32000) {
+ if (aec->num_bands > 1) {
// H band gain
// average nlp over low band: average over second half of freq spectrum
@@ -1047,19 +1049,21 @@
}
// compute gain factor
- for (i = 0; i < PART_LEN; i++) {
- dtmp = aec->dBufH[i];
- dtmp = dtmp * nlpGainHband; // for variable gain
+ for (j = 0; j < aec->num_bands - 1; ++j) {
+ for (i = 0; i < PART_LEN; i++) {
+ dtmp = aec->dBufH[j][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];
+ // add some comfort noise where Hband is attenuated
+ if (flagHbandCn == 1 && j == 0) {
+ fft[i] *= scale; // fft scaling
+ dtmp += cnScaleHband * fft[i];
+ }
+
+ // Saturate output to keep it in the allowed range.
+ outputH[j][i] = WEBRTC_SPL_SAT(
+ WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
}
-
- // Saturate output to keep it in the allowed range.
- outputH[i] = WEBRTC_SPL_SAT(
- WEBRTC_SPL_WORD16_MAX, dtmp, WEBRTC_SPL_WORD16_MIN);
}
}
@@ -1068,8 +1072,8 @@
memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
// Copy the current block to the old position for H band
- if (aec->sampFreq == 32000) {
- memcpy(aec->dBufH, aec->dBufH + PART_LEN, sizeof(float) * PART_LEN);
+ for (i = 0; i < aec->num_bands - 1; ++i) {
+ memcpy(aec->dBufH[i], aec->dBufH[i] + PART_LEN, sizeof(float) * PART_LEN);
}
memmove(aec->xfwBuf + PART_LEN1,
@@ -1101,14 +1105,21 @@
float nearend[PART_LEN];
float* nearend_ptr = NULL;
float output[PART_LEN];
- float outputH[PART_LEN];
+ float outputH[NUM_HIGH_BANDS_MAX][PART_LEN];
+ float* outputH_ptr[NUM_HIGH_BANDS_MAX];
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ outputH_ptr[i] = outputH[i];
+ }
float* xf_ptr = NULL;
// Concatenate old and new nearend blocks.
- if (aec->sampFreq == 32000) {
- WebRtc_ReadBuffer(aec->nearFrBufH, (void**)&nearend_ptr, nearend, PART_LEN);
- memcpy(aec->dBufH + PART_LEN, nearend_ptr, sizeof(nearend));
+ for (i = 0; i < aec->num_bands - 1; ++i) {
+ WebRtc_ReadBuffer(aec->nearFrBufH[i],
+ (void**)&nearend_ptr,
+ nearend,
+ PART_LEN);
+ memcpy(aec->dBufH[i] + 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));
@@ -1254,7 +1265,7 @@
// Scale error signal inversely with far power.
WebRtcAec_ScaleErrorSignal(aec, ef);
WebRtcAec_FilterAdaptation(aec, fft, ef);
- NonLinearProcessing(aec, output, outputH);
+ NonLinearProcessing(aec, output, outputH_ptr);
if (aec->metricsMode == 1) {
// Update power levels and echo metrics
@@ -1265,9 +1276,9 @@
// Store the output block.
WebRtc_WriteBuffer(aec->outFrBuf, output, PART_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtc_WriteBuffer(aec->outFrBufH, outputH, PART_LEN);
+ // For high bands
+ for (i = 0; i < aec->num_bands - 1; ++i) {
+ WebRtc_WriteBuffer(aec->outFrBufH[i], outputH[i], PART_LEN);
}
#ifdef WEBRTC_AEC_DEBUG_DUMP
@@ -1277,6 +1288,7 @@
}
int WebRtcAec_CreateAec(AecCore** aecInst) {
+ int i;
AecCore* aec = malloc(sizeof(AecCore));
*aecInst = aec;
if (aec == NULL) {
@@ -1297,18 +1309,21 @@
return -1;
}
- 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(float));
- if (!aec->outFrBufH) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ aec->nearFrBufH[i] = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(float));
+ if (!aec->nearFrBufH[i]) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
+ aec->outFrBufH[i] = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
+ sizeof(float));
+ if (!aec->outFrBufH[i]) {
+ WebRtcAec_FreeAec(aec);
+ aec = NULL;
+ return -1;
+ }
}
// Create far-end buffers.
@@ -1390,6 +1405,7 @@
}
int WebRtcAec_FreeAec(AecCore* aec) {
+ int i;
if (aec == NULL) {
return -1;
}
@@ -1397,8 +1413,10 @@
WebRtc_FreeBuffer(aec->nearFrBuf);
WebRtc_FreeBuffer(aec->outFrBuf);
- WebRtc_FreeBuffer(aec->nearFrBufH);
- WebRtc_FreeBuffer(aec->outFrBufH);
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ WebRtc_FreeBuffer(aec->nearFrBufH[i]);
+ WebRtc_FreeBuffer(aec->outFrBufH[i]);
+ }
WebRtc_FreeBuffer(aec->far_buf);
WebRtc_FreeBuffer(aec->far_buf_windowed);
@@ -1447,15 +1465,19 @@
if (sampFreq == 8000) {
aec->normal_mu = 0.6f;
aec->normal_error_threshold = 2e-6f;
+ aec->num_bands = 1;
} else {
aec->normal_mu = 0.5f;
aec->normal_error_threshold = 1.5e-6f;
+ aec->num_bands = sampFreq / 16000;
}
WebRtc_InitBuffer(aec->nearFrBuf);
WebRtc_InitBuffer(aec->outFrBuf);
- WebRtc_InitBuffer(aec->nearFrBufH);
- WebRtc_InitBuffer(aec->outFrBufH);
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ WebRtc_InitBuffer(aec->nearFrBufH[i]);
+ WebRtc_InitBuffer(aec->outFrBufH[i]);
+ }
// Initialize far-end buffers.
WebRtc_InitBuffer(aec->far_buf);
@@ -1516,7 +1538,7 @@
// Sampling frequency multiplier
// SWB is processed as 160 frame size
- if (aec->sampFreq == 32000) {
+ if (aec->num_bands > 1) {
aec->mult = (short)aec->sampFreq / 16000;
} else {
aec->mult = (short)aec->sampFreq / 8000;
@@ -1532,8 +1554,10 @@
// Initialize buffers
memset(aec->dBuf, 0, sizeof(aec->dBuf));
memset(aec->eBuf, 0, sizeof(aec->eBuf));
- // For H band
- memset(aec->dBufH, 0, sizeof(aec->dBufH));
+ // For H bands
+ for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
+ memset(aec->dBufH[i], 0, sizeof(aec->dBufH[i]));
+ }
memset(aec->xPow, 0, sizeof(aec->xPow));
memset(aec->dPow, 0, sizeof(aec->dPow));
@@ -1620,12 +1644,13 @@
return elements_moved;
}
-void WebRtcAec_ProcessFrame(AecCore* aec,
- const float* nearend,
- const float* nearendH,
- int knownDelay,
- float* out,
- float* outH) {
+void WebRtcAec_ProcessFrames(AecCore* aec,
+ const float* const* nearend,
+ int num_bands,
+ int num_samples,
+ int knownDelay,
+ float* const* out) {
+ int i, j;
int out_elements = 0;
// For each frame the process is as follows:
@@ -1655,94 +1680,100 @@
// Note that the two algorithms operate independently. Currently, we only
// allow one algorithm to be turned on.
- // TODO(bjornv): Change the near-end buffer handling to be the same as for
- // far-end, that is, with a near_pre_buf.
- // Buffer the near-end frame.
- WebRtc_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtc_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
- }
+ assert(aec->num_bands == num_bands);
- // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
- // have enough far-end data for that by stuffing the buffer if the
- // |system_delay| indicates others.
- if (aec->system_delay < FRAME_LEN) {
- // We don't have enough data so we rewind 10 ms.
- WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
- }
+ for (j = 0; j < num_samples; j+= FRAME_LEN) {
+ // TODO(bjornv): Change the near-end buffer handling to be the same as for
+ // far-end, that is, with a near_pre_buf.
+ // Buffer the near-end frame.
+ WebRtc_WriteBuffer(aec->nearFrBuf, &nearend[0][j], FRAME_LEN);
+ // For H band
+ for (i = 1; i < num_bands; ++i) {
+ WebRtc_WriteBuffer(aec->nearFrBufH[i - 1], &nearend[i][j], FRAME_LEN);
+ }
- if (aec->reported_delay_enabled) {
- // 2 a) Compensate for a possible change in the system delay.
-
- // TODO(bjornv): Investigate how we should round the delay difference; right
- // now we know that incoming |knownDelay| is underestimated when it's less
- // than |aec->knownDelay|. We therefore, round (-32) in that direction. In
- // the other direction, we don't have this situation, but might flush one
- // partition too little. This can cause non-causality, which should be
- // investigated. Maybe, allow for a non-symmetric rounding, like -16.
- int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
- int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
- WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
- aec->knownDelay -= moved_elements * PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
-#endif
- } else {
- // 2 b) Apply signal based delay correction.
- int move_elements = SignalBasedDelayCorrection(aec);
- int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
- WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
-#endif
- WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
- WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
- moved_elements);
- aec->signal_delay_correction += moved_elements;
- // TODO(bjornv): Investigate if this is reasonable. I had to add this
- // guard when the signal based delay correction replaces the system based
- // one. Otherwise there was a buffer underrun in the "qa-new/01/" recording
- // when adding 44 ms extra delay. This was not seen if we kept both delay
- // correction algorithms running in parallel.
- // A first investigation showed that we have a drift in this case that
- // causes the buffer underrun. Compared to when delay correction was
- // turned off, we get buffer underrun as well which was triggered in 1)
- // above. In addition there was a shift in |knownDelay| later increasing
- // the buffer. When running in parallel, this if statement was not
- // triggered. This suggests two alternatives; (a) use both algorithms, or
- // (b) allow for smaller delay corrections when we operate close to the
- // buffer limit. At the time of testing we required a change of 6 blocks,
- // but could change it to, e.g., 2 blocks. It requires some testing though.
- if ((int)WebRtc_available_read(aec->far_buf) < (aec->mult + 1)) {
- // We don't have enough data so we stuff the far-end buffers.
+ // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
+ // have enough far-end data for that by stuffing the buffer if the
+ // |system_delay| indicates others.
+ if (aec->system_delay < FRAME_LEN) {
+ // We don't have enough data so we rewind 10 ms.
WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
}
- }
- // 4) Process as many blocks as possible.
- while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
- ProcessBlock(aec);
- }
+ if (aec->reported_delay_enabled) {
+ // 2 a) Compensate for a possible change in the system delay.
- // 5) Update system delay with respect to the entire frame.
- aec->system_delay -= FRAME_LEN;
-
- // 6) Update output frame.
- // Stuff the out buffer if we have less than a frame to output.
- // This should only happen for the first frame.
- out_elements = (int)WebRtc_available_read(aec->outFrBuf);
- if (out_elements < FRAME_LEN) {
- WebRtc_MoveReadPtr(aec->outFrBuf, out_elements - FRAME_LEN);
- if (aec->sampFreq == 32000) {
- WebRtc_MoveReadPtr(aec->outFrBufH, out_elements - FRAME_LEN);
+ // TODO(bjornv): Investigate how we should round the delay difference;
+ // right now we know that incoming |knownDelay| is underestimated when
+ // it's less than |aec->knownDelay|. We therefore, round (-32) in that
+ // direction. In the other direction, we don't have this situation, but
+ // might flush one partition too little. This can cause non-causality,
+ // which should be investigated. Maybe, allow for a non-symmetric
+ // rounding, like -16.
+ int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
+ int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
+ aec->knownDelay -= moved_elements * PART_LEN;
+ #ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+ #endif
+ } else {
+ // 2 b) Apply signal based delay correction.
+ int move_elements = SignalBasedDelayCorrection(aec);
+ int moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
+ WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
+ #ifdef WEBRTC_AEC_DEBUG_DUMP
+ WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
+ #endif
+ WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements);
+ WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend,
+ moved_elements);
+ aec->signal_delay_correction += moved_elements;
+ // TODO(bjornv): Investigate if this is reasonable. I had to add this
+ // guard when the signal based delay correction replaces the system based
+ // one. Otherwise there was a buffer underrun in the "qa-new/01/"
+ // recording when adding 44 ms extra delay. This was not seen if we kept
+ // both delay correction algorithms running in parallel.
+ // A first investigation showed that we have a drift in this case that
+ // causes the buffer underrun. Compared to when delay correction was
+ // turned off, we get buffer underrun as well which was triggered in 1)
+ // above. In addition there was a shift in |knownDelay| later increasing
+ // the buffer. When running in parallel, this if statement was not
+ // triggered. This suggests two alternatives; (a) use both algorithms, or
+ // (b) allow for smaller delay corrections when we operate close to the
+ // buffer limit. At the time of testing we required a change of 6 blocks,
+ // but could change it to, e.g., 2 blocks. It requires some testing
+ // though.
+ if ((int)WebRtc_available_read(aec->far_buf) < (aec->mult + 1)) {
+ // We don't have enough data so we stuff the far-end buffers.
+ WebRtcAec_MoveFarReadPtr(aec, -(aec->mult + 1));
+ }
}
- }
- // Obtain an output frame.
- WebRtc_ReadBuffer(aec->outFrBuf, NULL, out, FRAME_LEN);
- // For H band.
- if (aec->sampFreq == 32000) {
- WebRtc_ReadBuffer(aec->outFrBufH, NULL, outH, FRAME_LEN);
+
+ // 4) Process as many blocks as possible.
+ while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
+ ProcessBlock(aec);
+ }
+
+ // 5) Update system delay with respect to the entire frame.
+ aec->system_delay -= FRAME_LEN;
+
+ // 6) Update output frame.
+ // Stuff the out buffer if we have less than a frame to output.
+ // This should only happen for the first frame.
+ out_elements = (int)WebRtc_available_read(aec->outFrBuf);
+ if (out_elements < FRAME_LEN) {
+ WebRtc_MoveReadPtr(aec->outFrBuf, out_elements - FRAME_LEN);
+ for (i = 0; i < num_bands - 1; ++i) {
+ WebRtc_MoveReadPtr(aec->outFrBufH[i], out_elements - FRAME_LEN);
+ }
+ }
+ // Obtain an output frame.
+ WebRtc_ReadBuffer(aec->outFrBuf, NULL, &out[0][j], FRAME_LEN);
+ // For H bands.
+ for (i = 1; i < num_bands; ++i) {
+ WebRtc_ReadBuffer(aec->outFrBufH[i - 1], NULL, &out[i][j], FRAME_LEN);
+ }
}
}
diff --git a/webrtc/modules/audio_processing/aec/aec_core.h b/webrtc/modules/audio_processing/aec/aec_core.h
index 93bfed4..2bffeec 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/webrtc/modules/audio_processing/aec/aec_core.h
@@ -21,6 +21,7 @@
#define PART_LEN 64 // Length of partition
#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
+#define NUM_HIGH_BANDS_MAX 2 // Max number of high bands
typedef float complex_t[2];
// For performance reasons, some arrays of complex numbers are replaced by twice
@@ -62,12 +63,12 @@
#endif
void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend);
-void WebRtcAec_ProcessFrame(AecCore* aec,
- const float* nearend,
- const float* nearendH,
- int knownDelay,
- float* out,
- float* outH);
+void WebRtcAec_ProcessFrames(AecCore* aec,
+ const float* const* nearend,
+ int num_bands,
+ int num_samples,
+ int knownDelay,
+ float* const* out);
// A helper function to call WebRtc_MoveReadPtr() for all far-end buffers.
// Returns the number of elements moved, and adjusts |system_delay| by the
diff --git a/webrtc/modules/audio_processing/aec/aec_core_internal.h b/webrtc/modules/audio_processing/aec/aec_core_internal.h
index fd33acd..2081813 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -59,13 +59,13 @@
RingBuffer* nearFrBuf;
RingBuffer* outFrBuf;
- RingBuffer* nearFrBufH;
- RingBuffer* outFrBufH;
+ RingBuffer* nearFrBufH[NUM_HIGH_BANDS_MAX];
+ RingBuffer* outFrBufH[NUM_HIGH_BANDS_MAX];
float dBuf[PART_LEN2]; // nearend
float eBuf[PART_LEN2]; // error
- float dBufH[PART_LEN2]; // nearend
+ float dBufH[NUM_HIGH_BANDS_MAX][PART_LEN2]; // nearend
float xPow[PART_LEN1];
float dPow[PART_LEN1];
@@ -101,6 +101,7 @@
int mult; // sampling frequency multiple
int sampFreq;
+ int num_bands;
uint32_t seed;
float normal_mu; // stepsize
diff --git a/webrtc/modules/audio_processing/aec/aec_core_mips.c b/webrtc/modules/audio_processing/aec/aec_core_mips.c
index 0a975f7..bb33087 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_mips.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_mips.c
@@ -274,7 +274,7 @@
noiseAvg = 0.0;
tmpAvg = 0.0;
num = 0;
- if (aec->sampFreq == 32000 && flagHbandCn == 1) {
+ if ((aec->sampFreq == 32000 || aec->sampFreq == 48000) && flagHbandCn == 1) {
for (i = 0; i < PART_LEN; i++) {
rand[i] = ((float)randW16[i]) / 32768;
}
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 99883c4..2cff8a2 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -104,18 +104,16 @@
static void EstBufDelayNormal(Aec* aecInst);
static void EstBufDelayExtended(Aec* aecInst);
static int ProcessNormal(Aec* self,
- const float* near,
- const float* near_high,
- float* out,
- float* out_high,
+ const float* const* near,
+ int num_bands,
+ float* const* out,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
static void ProcessExtended(Aec* self,
- const float* near,
- const float* near_high,
- float* out,
- float* out_high,
+ const float* const* near,
+ int num_bands,
+ float* const* out,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
@@ -199,7 +197,10 @@
Aec* aecpc = aecInst;
AecConfig aecConfig;
- if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
+ if (sampFreq != 8000 &&
+ sampFreq != 16000 &&
+ sampFreq != 32000 &&
+ sampFreq != 48000) {
aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
return -1;
}
@@ -227,7 +228,7 @@
aecpc->initFlag = initCheck; // indicates that initialization has been done
- if (aecpc->sampFreq == 32000) {
+ if (aecpc->sampFreq == 32000 || aecpc->sampFreq == 48000) {
aecpc->splitSampFreq = 16000;
} else {
aecpc->splitSampFreq = sampFreq;
@@ -338,19 +339,14 @@
}
int32_t WebRtcAec_Process(void* aecInst,
- const float* nearend,
- const float* nearendH,
- float* out,
- float* outH,
+ const float* const* nearend,
+ int num_bands,
+ float* const* out,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew) {
Aec* aecpc = aecInst;
int32_t retVal = 0;
- if (nearend == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
if (out == NULL) {
aecpc->lastError = AEC_NULL_POINTER_ERROR;
@@ -368,12 +364,6 @@
return -1;
}
- // Check for valid pointers based on sampling rate
- if (aecpc->sampFreq == 32000 && nearendH == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
if (msInSndCardBuf < 0) {
msInSndCardBuf = 0;
aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
@@ -386,14 +376,18 @@
// This returns the value of aec->extended_filter_enabled.
if (WebRtcAec_delay_correction_enabled(aecpc->aec)) {
- ProcessExtended(
- aecpc, nearend, nearendH, out, outH, nrOfSamples, msInSndCardBuf, skew);
+ ProcessExtended(aecpc,
+ nearend,
+ num_bands,
+ out,
+ nrOfSamples,
+ msInSndCardBuf,
+ skew);
} else {
if (ProcessNormal(aecpc,
nearend,
- nearendH,
+ num_bands,
out,
- outH,
nrOfSamples,
msInSndCardBuf,
skew) != 0) {
@@ -598,17 +592,15 @@
}
static int ProcessNormal(Aec* aecpc,
- const float* nearend,
- const float* nearendH,
- float* out,
- float* outH,
+ const float* const* nearend,
+ int num_bands,
+ float* const* out,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew) {
int retVal = 0;
short i;
short nBlocks10ms;
- short nFrames;
// Limit resampling to doubling/halving of signal
const float minSkewEst = -0.5f;
const float maxSkewEst = 1.0f;
@@ -649,16 +641,14 @@
}
}
- nFrames = nrOfSamples / FRAME_LEN;
- nBlocks10ms = nFrames / aecpc->rate_factor;
+ nBlocks10ms = nrOfSamples / (FRAME_LEN * aecpc->rate_factor);
if (aecpc->startup_phase) {
- // Only needed if they don't already point to the same place.
- if (nearend != out) {
- memcpy(out, nearend, sizeof(*out) * nrOfSamples);
- }
- if (nearendH != outH) {
- memcpy(outH, nearendH, sizeof(*outH) * nrOfSamples);
+ for (i = 0; i < num_bands; ++i) {
+ // Only needed if they don't already point to the same place.
+ if (nearend[i] != out[i]) {
+ memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * nrOfSamples);
+ }
}
// The AEC is in the start up mode
@@ -736,34 +726,29 @@
EstBufDelayNormal(aecpc);
}
- // Note that 1 frame is supported for NB and 2 frames for WB.
- for (i = 0; i < nFrames; i++) {
- // Call the AEC.
- WebRtcAec_ProcessFrame(aecpc->aec,
- &nearend[FRAME_LEN * i],
- &nearendH[FRAME_LEN * i],
- aecpc->knownDelay,
- &out[FRAME_LEN * i],
- &outH[FRAME_LEN * i]);
- // TODO(bjornv): Re-structure such that we don't have to pass
- // |aecpc->knownDelay| as input. Change name to something like
- // |system_buffer_diff|.
- }
+ // Call the AEC.
+ // TODO(bjornv): Re-structure such that we don't have to pass
+ // |aecpc->knownDelay| as input. Change name to something like
+ // |system_buffer_diff|.
+ WebRtcAec_ProcessFrames(aecpc->aec,
+ nearend,
+ num_bands,
+ nrOfSamples,
+ aecpc->knownDelay,
+ out);
}
return retVal;
}
static void ProcessExtended(Aec* self,
- const float* near,
- const float* near_high,
- float* out,
- float* out_high,
+ const float* const* near,
+ int num_bands,
+ float* const* out,
int16_t num_samples,
int16_t reported_delay_ms,
int32_t skew) {
int i;
- const int num_frames = num_samples / FRAME_LEN;
const int delay_diff_offset = kDelayDiffOffsetSamples;
#if defined(WEBRTC_UNTRUSTED_DELAY)
reported_delay_ms = kFixedDelayMs;
@@ -786,12 +771,11 @@
self->msInSndCardBuf = reported_delay_ms;
if (!self->farend_started) {
- // Only needed if they don't already point to the same place.
- if (near != out) {
- memcpy(out, near, sizeof(*out) * num_samples);
- }
- if (near_high != out_high) {
- memcpy(out_high, near_high, sizeof(*out_high) * num_samples);
+ for (i = 0; i < num_bands; ++i) {
+ // Only needed if they don't already point to the same place.
+ if (near[i] != out[i]) {
+ memcpy(out[i], near[i], sizeof(near[i][0]) * num_samples);
+ }
}
return;
}
@@ -820,14 +804,12 @@
const int adjusted_known_delay =
WEBRTC_SPL_MAX(0, self->knownDelay + delay_diff_offset);
- for (i = 0; i < num_frames; ++i) {
- WebRtcAec_ProcessFrame(self->aec,
- &near[FRAME_LEN * i],
- &near_high[FRAME_LEN * i],
- adjusted_known_delay,
- &out[FRAME_LEN * i],
- &out_high[FRAME_LEN * i]);
- }
+ WebRtcAec_ProcessFrames(self->aec,
+ near,
+ num_bands,
+ num_samples,
+ adjusted_known_delay,
+ out);
}
}
diff --git a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
index 0cf6a5a..51eb37a 100644
--- a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
+++ b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
@@ -133,10 +133,9 @@
* Inputs Description
* -------------------------------------------------------------------
* void* aecInst Pointer to the AEC instance
- * float* nearend In buffer containing one frame of
- * nearend+echo signal for L band
- * float* nearendH In buffer containing one frame of
- * nearend+echo signal for H band
+ * float* const* nearend In buffer containing one frame of
+ * nearend+echo signal for each band
+ * int num_bands Number of bands in nearend buffer
* int16_t nrOfSamples Number of samples in nearend buffer
* int16_t msInSndCardBuf Delay estimate for sound card and
* system buffers
@@ -146,18 +145,15 @@
*
* Outputs Description
* -------------------------------------------------------------------
- * float* out Out buffer, one frame of processed nearend
- * for L band
- * float* outH Out buffer, one frame of processed nearend
- * for H band
+ * float* const* out Out buffer, one frame of processed nearend
+ * for each band
* int32_t return 0: OK
* -1: error
*/
int32_t WebRtcAec_Process(void* aecInst,
- const float* nearend,
- const float* nearendH,
- float* out,
- float* outH,
+ const float* const* nearend,
+ int num_bands,
+ float* const* out,
int16_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew);
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index 927c317..654ae54 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -50,6 +50,8 @@
float far_[kSamplesPerChunk];
float near_[kSamplesPerChunk];
float out_[kSamplesPerChunk];
+ const float* near_ptr_;
+ float* out_ptr_;
};
SystemDelayTest::SystemDelayTest()
@@ -60,6 +62,8 @@
near_[i] = 514.0;
}
memset(out_, 0, sizeof(out_));
+ near_ptr_ = near_;
+ out_ptr_ = out_;
}
void SystemDelayTest::SetUp() {
@@ -103,10 +107,9 @@
EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
EXPECT_EQ(0,
WebRtcAec_Process(handle_,
- near_,
- NULL,
- out_,
- NULL,
+ &near_ptr_,
+ 1,
+ &out_ptr_,
samples_per_frame_,
device_buffer_ms,
0));
@@ -268,10 +271,9 @@
for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
EXPECT_EQ(0,
WebRtcAec_Process(handle_,
- near_,
- NULL,
- out_,
- NULL,
+ &near_ptr_,
+ 1,
+ &out_ptr_,
samples_per_frame_,
kDeviceBufMs,
0));
@@ -322,10 +324,9 @@
for (int j = 0; j <= kStableConvergenceMs; j += 10) {
EXPECT_EQ(0,
WebRtcAec_Process(handle_,
- near_,
- NULL,
- out_,
- NULL,
+ &near_ptr_,
+ 1,
+ &out_ptr_,
samples_per_frame_,
kDeviceBufMs,
0));
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index 863f2d8..bf5ed72 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -129,10 +129,9 @@
Handle* my_handle = handle(handle_index);
err = WebRtcAec_Process(
my_handle,
- audio->split_bands_const_f(i)[kBand0To8kHz],
- audio->split_bands_const_f(i)[kBand8To16kHz],
- audio->split_bands_f(i)[kBand0To8kHz],
- audio->split_bands_f(i)[kBand8To16kHz],
+ audio->split_bands_const_f(i),
+ audio->num_bands(),
+ audio->split_bands_f(i),
static_cast<int16_t>(audio->samples_per_split_channel()),
apm_->stream_delay_ms(),
stream_drift_samples_);
