Index: webrtc/modules/audio_processing/aec/aec_core.c |
diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c |
index cf72e8390de08b461cc91d960889d589232b5de8..013dc78e70a5e8ddaaadabbfeb972972217a90db 100644 |
--- a/webrtc/modules/audio_processing/aec/aec_core.c |
+++ b/webrtc/modules/audio_processing/aec/aec_core.c |
@@ -842,11 +842,7 @@ static void Fft(float time_data[PART_LEN2], |
} |
static int MoveFarReadPtrWithoutSystemDelayUpdate(AecCore* self, int elements) { |
peah-webrtc
2015/12/16 12:52:22
I think that since this is now a one-liner, we sho
minyue-webrtc
2015/12/16 14:47:17
yes, certainly.
|
- WebRtc_MoveReadPtr(self->far_buf_windowed, elements); |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
- WebRtc_MoveReadPtr(self->far_time_buf, elements); |
-#endif |
- return WebRtc_MoveReadPtr(self->far_buf, elements); |
+ return WebRtc_MoveReadPtr(self->far_time_buf, elements); |
} |
static int SignalBasedDelayCorrection(AecCore* self) { |
@@ -888,7 +884,7 @@ static int SignalBasedDelayCorrection(AecCore* self) { |
const int upper_bound = self->num_partitions * 3 / 4; |
const int do_correction = delay <= lower_bound || delay > upper_bound; |
if (do_correction == 1) { |
- int available_read = (int)WebRtc_available_read(self->far_buf); |
+ int available_read = (int)WebRtc_available_read(self->far_time_buf); |
// With |shift_offset| we gradually rely on the delay estimates. For |
// positive delays we reduce the correction by |shift_offset| to lower the |
// risk of pushing the AEC into a non causal state. For negative delays |
@@ -1003,6 +999,7 @@ static void EchoSubtraction( |
static void EchoSuppression(AecCore* aec, |
+ float farend[PART_LEN2], |
float* echo_subtractor_output, |
float* output, |
float* const* outputH) { |
@@ -1056,13 +1053,13 @@ static void EchoSuppression(AecCore* aec, |
aec->delayEstCtr = 0; |
} |
- // We should always have at least one element stored in |far_buf|. |
- assert(WebRtc_available_read(aec->far_buf_windowed) > 0); |
// NLP |
- WebRtc_ReadBuffer(aec->far_buf_windowed, (void**)&xfw_ptr, &xfw[0][0], 1); |
- // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of |
- // |xfwBuf|. |
+ // Convert far-end partition to the frequency domain with windowing. |
+ WindowData(fft, farend); |
+ Fft(fft, xfw); |
+ xfw_ptr = &xfw[0][0]; |
+ |
// Buffer far. |
memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1); |
@@ -1285,8 +1282,10 @@ static void ProcessBlock(AecCore* aec) { |
const float gInitNoise[2] = {0.999f, 0.001f}; |
float nearend[PART_LEN]; |
- float echo_subtractor_output[PART_LEN]; |
float* nearend_ptr = NULL; |
+ float farend[PART_LEN2]; |
+ float* farend_ptr = NULL; |
+ float echo_subtractor_output[PART_LEN]; |
float output[PART_LEN]; |
float outputH[NUM_HIGH_BANDS_MAX][PART_LEN]; |
float* outputH_ptr[NUM_HIGH_BANDS_MAX]; |
@@ -1307,21 +1306,24 @@ static void ProcessBlock(AecCore* aec) { |
WebRtc_ReadBuffer(aec->nearFrBuf, (void**)&nearend_ptr, nearend, PART_LEN); |
memcpy(aec->dBuf + PART_LEN, nearend_ptr, sizeof(nearend)); |
- // ---------- Ooura fft ---------- |
+ // We should always have at least one element stored in |far_buf|. |
+ assert(WebRtc_available_read(aec->far_time_buf) > 0); |
+ WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1); |
#ifdef WEBRTC_AEC_DEBUG_DUMP |
{ |
- float farend[PART_LEN]; |
- float* farend_ptr = NULL; |
- WebRtc_ReadBuffer(aec->far_time_buf, (void**)&farend_ptr, farend, 1); |
- RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, farend_ptr, PART_LEN); |
+ // TODO(minyue): |farend_ptr| starts from buffered samples. This will be |
+ // modified when |aec->far_time_buf| is revised. |
+ RTC_AEC_DEBUG_WAV_WRITE(aec->farFile, &farend_ptr[PART_LEN], PART_LEN); |
+ |
RTC_AEC_DEBUG_WAV_WRITE(aec->nearFile, nearend_ptr, PART_LEN); |
} |
#endif |
- // We should always have at least one element stored in |far_buf|. |
- assert(WebRtc_available_read(aec->far_buf) > 0); |
- WebRtc_ReadBuffer(aec->far_buf, (void**)&xf_ptr, &xf[0][0], 1); |
+ // Convert far-end signal to the frequency domain. |
+ memcpy(fft, farend_ptr, sizeof(float) * PART_LEN2); |
+ Fft(fft, xf); |
+ xf_ptr = &xf[0][0]; |
// Near fft |
memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2); |
@@ -1421,7 +1423,7 @@ static void ProcessBlock(AecCore* aec) { |
RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN); |
// Perform echo suppression. |
- EchoSuppression(aec, echo_subtractor_output, output, outputH_ptr); |
+ EchoSuppression(aec, farend_ptr, echo_subtractor_output, output, outputH_ptr); |
if (aec->metricsMode == 1) { |
// Update power levels and echo metrics |
@@ -1475,26 +1477,20 @@ AecCore* WebRtcAec_CreateAec() { |
} |
// Create far-end buffers. |
- aec->far_buf = |
- WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1); |
- if (!aec->far_buf) { |
- WebRtcAec_FreeAec(aec); |
- return NULL; |
- } |
- aec->far_buf_windowed = |
- WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1); |
- if (!aec->far_buf_windowed) { |
- WebRtcAec_FreeAec(aec); |
- return NULL; |
- } |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
- aec->instance_index = webrtc_aec_instance_count; |
+ // For bit exactness with legacy code, each element in |far_time_buf| is |
+ // supposed to contain |PART_LEN2| samples with an overlap of |PART_LEN| |
+ // samples from the last frame. |
+ // TODO(minyue): reduce |far_time_buf| to non-overlapped |PART_LEN| samples. |
aec->far_time_buf = |
- WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN); |
+ WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN2); |
if (!aec->far_time_buf) { |
WebRtcAec_FreeAec(aec); |
return NULL; |
} |
+ |
+#ifdef WEBRTC_AEC_DEBUG_DUMP |
+ aec->instance_index = webrtc_aec_instance_count; |
+ |
aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL; |
aec->debug_dump_count = 0; |
#endif |
@@ -1572,11 +1568,8 @@ void WebRtcAec_FreeAec(AecCore* aec) { |
WebRtc_FreeBuffer(aec->outFrBufH[i]); |
} |
- WebRtc_FreeBuffer(aec->far_buf); |
- WebRtc_FreeBuffer(aec->far_buf_windowed); |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
WebRtc_FreeBuffer(aec->far_time_buf); |
-#endif |
+ |
RTC_AEC_DEBUG_WAV_CLOSE(aec->farFile); |
RTC_AEC_DEBUG_WAV_CLOSE(aec->nearFile); |
RTC_AEC_DEBUG_WAV_CLOSE(aec->outFile); |
@@ -1612,10 +1605,9 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { |
} |
// Initialize far-end buffers. |
- WebRtc_InitBuffer(aec->far_buf); |
- WebRtc_InitBuffer(aec->far_buf_windowed); |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
WebRtc_InitBuffer(aec->far_time_buf); |
+ |
+#ifdef WEBRTC_AEC_DEBUG_DUMP |
{ |
int process_rate = sampFreq > 16000 ? 16000 : sampFreq; |
RTC_AEC_DEBUG_WAV_REOPEN("aec_far", aec->instance_index, |
@@ -1759,23 +1751,18 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) { |
return 0; |
} |
-void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) { |
- float fft[PART_LEN2]; |
- float xf[2][PART_LEN1]; |
+// For bit exactness with a legacy code, |farend| is supposed to contain |
+// |PART_LEN2| samples with an overlap of |PART_LEN| samples from the last |
+// frame. |
+// TODO(minyue): reduce |farend| to non-overlapped |PART_LEN| samples. |
+void WebRtcAec_BufferFarendPartition(AecCore* aec, const float* farend) { |
// Check if the buffer is full, and in that case flush the oldest data. |
- if (WebRtc_available_write(aec->far_buf) < 1) { |
+ if (WebRtc_available_write(aec->far_time_buf) < 1) { |
WebRtcAec_MoveFarReadPtr(aec, 1); |
} |
- // Convert far-end partition to the frequency domain without windowing. |
- memcpy(fft, farend, sizeof(float) * PART_LEN2); |
- Fft(fft, xf); |
- WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1); |
- // Convert far-end partition to the frequency domain with windowing. |
- WindowData(fft, farend); |
- Fft(fft, xf); |
- WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1); |
+ WebRtc_WriteBuffer(aec->far_time_buf, farend, 1); |
} |
int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements) { |
@@ -1860,7 +1847,7 @@ void WebRtcAec_ProcessFrames(AecCore* aec, |
int move_elements = SignalBasedDelayCorrection(aec); |
int moved_elements = |
MoveFarReadPtrWithoutSystemDelayUpdate(aec, move_elements); |
- int far_near_buffer_diff = WebRtc_available_read(aec->far_buf) - |
+ int far_near_buffer_diff = WebRtc_available_read(aec->far_time_buf) - |
WebRtc_available_read(aec->nearFrBuf) / PART_LEN; |
WebRtc_SoftResetDelayEstimator(aec->delay_estimator, moved_elements); |
WebRtc_SoftResetDelayEstimatorFarend(aec->delay_estimator_farend, |
@@ -1939,10 +1926,6 @@ void WebRtcAec_GetEchoStats(AecCore* self, |
*a_nlp = self->aNlp; |
} |
-#ifdef WEBRTC_AEC_DEBUG_DUMP |
-void* WebRtcAec_far_time_buf(AecCore* self) { return self->far_time_buf; } |
-#endif |
- |
void WebRtcAec_SetConfigCore(AecCore* self, |
int nlp_mode, |
int metrics_mode, |