EchoCancellationImpl::ProcessRenderAudio: Use float samples directly
This patch lets EchoCancellationImpl::ProcessRenderAudio ask the given
AudioBuffer for float sample data directly, instead of asking for
int16 samples and then converting manually.
Since EchoCancellationImpl::ProcessRenderAudio takes a const
AudioBuffer*, it was necessary to add some const accessors for float
data to AudioBuffer.
R=aluebs@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6590 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.c b/webrtc/modules/audio_processing/aec/aec_resampler.c
index 5382665..469b811 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.c
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.c
@@ -26,7 +26,7 @@
};
typedef struct {
- short buffer[kResamplerBufferSize];
+ float buffer[kResamplerBufferSize];
float position;
int deviceSampleRateHz;
@@ -71,15 +71,15 @@
}
void WebRtcAec_ResampleLinear(void* resampInst,
- const short* inspeech,
+ const float* inspeech,
int size,
float skew,
- short* outspeech,
+ float* outspeech,
int* size_out) {
resampler_t* obj = (resampler_t*)resampInst;
- short* y;
- float be, tnew, interp;
+ float* y;
+ float be, tnew;
int tn, mm;
assert(!(size < 0 || size > 2 * FRAME_LEN));
@@ -91,7 +91,7 @@
// Add new frame data in lookahead
memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
inspeech,
- size * sizeof(short));
+ size * sizeof(inspeech[0]));
// Sample rate ratio
be = 1 + skew;
@@ -106,15 +106,7 @@
while (tn < size) {
// Interpolation
- interp = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]);
-
- if (interp > 32767) {
- interp = 32767;
- } else if (interp < -32768) {
- interp = -32768;
- }
-
- outspeech[mm] = (short)interp;
+ outspeech[mm] = y[tn] + (tnew - tn) * (y[tn + 1] - y[tn]);
mm++;
tnew = be * mm + obj->position;
@@ -127,7 +119,7 @@
// Shift buffer
memmove(obj->buffer,
&obj->buffer[size],
- (kResamplerBufferSize - size) * sizeof(short));
+ (kResamplerBufferSize - size) * sizeof(obj->buffer[0]));
}
int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst) {
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.h b/webrtc/modules/audio_processing/aec/aec_resampler.h
index e42c056..73e2821 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.h
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.h
@@ -30,10 +30,10 @@
// Resamples input using linear interpolation.
void WebRtcAec_ResampleLinear(void* resampInst,
- const short* inspeech,
+ const float* inspeech,
int size,
float skew,
- short* outspeech,
+ float* outspeech,
int* size_out);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index ba3b924..b58edcb 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -294,17 +294,12 @@
// only buffer L band for farend
int32_t WebRtcAec_BufferFarend(void* aecInst,
- const int16_t* farend,
+ const float* farend,
int16_t nrOfSamples) {
aecpc_t* aecpc = aecInst;
- int32_t retVal = 0;
int newNrOfSamples = (int)nrOfSamples;
- short newFarend[MAX_RESAMP_LEN];
- const int16_t* farend_ptr = farend;
- float tmp_farend[MAX_RESAMP_LEN];
- const float* farend_float = tmp_farend;
- float skew;
- int i = 0;
+ float new_farend[MAX_RESAMP_LEN];
+ const float* farend_ptr = farend;
if (farend == NULL) {
aecpc->lastError = AEC_NULL_POINTER_ERROR;
@@ -322,17 +317,15 @@
return -1;
}
- skew = aecpc->skew;
-
if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
// Resample and get a new number of samples
WebRtcAec_ResampleLinear(aecpc->resampler,
farend,
nrOfSamples,
- skew,
- newFarend,
+ aecpc->skew,
+ new_farend,
&newNrOfSamples);
- farend_ptr = (const int16_t*)newFarend;
+ farend_ptr = new_farend;
}
aecpc->farend_started = 1;
@@ -343,32 +336,31 @@
WebRtc_WriteBuffer(
aecpc->far_pre_buf_s16, farend_ptr, (size_t)newNrOfSamples);
#endif
- // Cast to float and write the time-domain data to |far_pre_buf|.
- for (i = 0; i < newNrOfSamples; i++) {
- tmp_farend[i] = (float)farend_ptr[i];
- }
- WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_float, (size_t)newNrOfSamples);
+ // Write the time-domain data to |far_pre_buf|.
+ WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, (size_t)newNrOfSamples);
// Transform to frequency domain if we have enough data.
while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
// We have enough data to pass to the FFT, hence read PART_LEN2 samples.
- WebRtc_ReadBuffer(
- aecpc->far_pre_buf, (void**)&farend_float, tmp_farend, PART_LEN2);
-
- WebRtcAec_BufferFarendPartition(aecpc->aec, farend_float);
+ {
+ float* ptmp;
+ float tmp[PART_LEN2];
+ WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**)&ptmp, tmp, PART_LEN2);
+ WebRtcAec_BufferFarendPartition(aecpc->aec, ptmp);
+ }
// Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN);
#ifdef WEBRTC_AEC_DEBUG_DUMP
WebRtc_ReadBuffer(
- aecpc->far_pre_buf_s16, (void**)&farend_ptr, newFarend, PART_LEN2);
+ aecpc->far_pre_buf_s16, (void**)&farend_ptr, new_farend, PART_LEN2);
WebRtc_WriteBuffer(
WebRtcAec_far_time_buf(aecpc->aec), &farend_ptr[PART_LEN], 1);
WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN);
#endif
}
- return retVal;
+ return 0;
}
int32_t WebRtcAec_Process(void* aecInst,
diff --git a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
index dc64a34..0cf6a5a 100644
--- a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
+++ b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
@@ -114,7 +114,7 @@
* Inputs Description
* -------------------------------------------------------------------
* void* aecInst Pointer to the AEC instance
- * int16_t* farend In buffer containing one frame of
+ * const float* farend In buffer containing one frame of
* farend signal for L band
* int16_t nrOfSamples Number of samples in farend buffer
*
@@ -124,7 +124,7 @@
* -1: error
*/
int32_t WebRtcAec_BufferFarend(void* aecInst,
- const int16_t* farend,
+ const float* farend,
int16_t nrOfSamples);
/*
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index 5fbc560..f81ce47 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -47,7 +47,7 @@
int samples_per_frame_;
// Dummy input/output speech data.
static const int kSamplesPerChunk = 160;
- int16_t far_[kSamplesPerChunk];
+ float far_[kSamplesPerChunk];
float near_[kSamplesPerChunk];
float out_[kSamplesPerChunk];
};
@@ -55,9 +55,10 @@
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_));
- for (int i = 0; i < kSamplesPerChunk; i++)
+ for (int i = 0; i < kSamplesPerChunk; i++) {
+ far_[i] = 257.0;
near_[i] = 514.0;
+ }
memset(out_, 0, sizeof(out_));
}
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index b0f1eb6..35e1eb7 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -294,11 +294,16 @@
return const_cast<int16_t*>(t->data(channel));
}
-float* AudioBuffer::data_f(int channel) {
+const float* AudioBuffer::data_f(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return channels_->fbuf()->channel(channel);
}
+float* AudioBuffer::data_f(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<float*>(t->data_f(channel));
+}
+
const int16_t* AudioBuffer::low_pass_split_data(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return split_channels_.get() ? split_channels_->low_channel(channel)
@@ -310,12 +315,17 @@
return const_cast<int16_t*>(t->low_pass_split_data(channel));
}
-float* AudioBuffer::low_pass_split_data_f(int channel) {
+const float* AudioBuffer::low_pass_split_data_f(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return split_channels_.get() ? split_channels_->low_channel_f(channel)
: data_f(channel);
}
+float* AudioBuffer::low_pass_split_data_f(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<float*>(t->low_pass_split_data_f(channel));
+}
+
const int16_t* AudioBuffer::high_pass_split_data(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return split_channels_.get() ? split_channels_->high_channel(channel) : NULL;
@@ -326,12 +336,17 @@
return const_cast<int16_t*>(t->high_pass_split_data(channel));
}
-float* AudioBuffer::high_pass_split_data_f(int channel) {
+const float* AudioBuffer::high_pass_split_data_f(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return split_channels_.get() ? split_channels_->high_channel_f(channel)
: NULL;
}
+float* AudioBuffer::high_pass_split_data_f(int channel) {
+ const AudioBuffer* t = this;
+ return const_cast<float*>(t->high_pass_split_data_f(channel));
+}
+
const int16_t* AudioBuffer::mixed_data(int channel) const {
assert(channel >= 0 && channel < num_mixed_channels_);
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 67e4f48..2fab814 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -69,8 +69,11 @@
// Float versions of the accessors, with automatic conversion back and forth
// as necessary. The range of the numbers are the same as for int16_t.
float* data_f(int channel);
+ const float* data_f(int channel) const;
float* low_pass_split_data_f(int channel);
+ const float* low_pass_split_data_f(int channel) const;
float* high_pass_split_data_f(int channel);
+ const float* high_pass_split_data_f(int channel) const;
const float* keyboard_data() const;
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index e770f9f..47b4f18 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -89,7 +89,7 @@
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
err = WebRtcAec_BufferFarend(
my_handle,
- audio->low_pass_split_data(j),
+ audio->low_pass_split_data_f(j),
static_cast<int16_t>(audio->samples_per_split_channel()));
if (err != apm_->kNoError) {
