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Using buffered signal to calculate the level of echo cancellation.

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The level of the error signal after linear echo cancellation was based on non-buffered signal while that of the near-end and far-end signal based on buffered signal. This discrepancy made the comparison of them unfair.

This CL is to make calculating the error level rely on the same buffering.

BUG=

Review URL: https://codereview.webrtc.org/1510873004

Cr-Commit-Position: refs/heads/master@{#11408}
This commit is contained in:
minyue 2016-01-27 15:44:52 -08:00 committed by Commit bot
parent d162a5e379
commit 691b8369ff
2 changed files with 17 additions and 25 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

38
webrtc/modules/audio_processing/aec/aec_core.c
View File

@ -667,12 +667,11 @@ static void UpdateMetrics(AecCore* aec) {
// A_NLP
dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
(2 * aec->linoutlevel.averagelevel) +
1e-10f);
aec->linoutlevel.averagelevel + 1e-10f);
// subtract noise power
suppressedEcho = 2 * (aec->linoutlevel.averagelevel -
safety * aec->linoutlevel.minlevel);
suppressedEcho = aec->linoutlevel.averagelevel -
safety * aec->linoutlevel.minlevel;
dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
@ -903,7 +902,6 @@ static void EchoSubtraction(
AecCore* aec,
int num_partitions,
int x_fft_buf_block_pos,
int metrics_mode,
int extended_filter_enabled,
float normal_mu,
float normal_error_threshold,
@ -911,7 +909,6 @@ static void EchoSubtraction(
float* const y,
float x_pow[PART_LEN1],
float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
PowerLevel* linout_level,
float echo_subtractor_output[PART_LEN]) {
float s_fft[2][PART_LEN1];
float e_extended[PART_LEN2];
@ -955,13 +952,6 @@ static void EchoSubtraction(
&e_fft[0][0],
sizeof(e_fft[0][0]) * PART_LEN1 * 2);
if (metrics_mode == 1) {
// Note that the first PART_LEN samples in fft (before transformation) are
// zero. Hence, the scaling by two in UpdateLevel() should not be
// performed. That scaling is taken care of in UpdateMetrics() instead.
UpdateLevel(linout_level, CalculatePower(e, PART_LEN) / 2.0f);
}
// Scale error signal inversely with far power.
WebRtcAec_ScaleErrorSignal(extended_filter_enabled,
normal_mu,
@ -976,7 +966,6 @@ static void EchoSubtraction(
memcpy(echo_subtractor_output, e, sizeof(float) * PART_LEN);
}
static void EchoSuppression(AecCore* aec,
float farend[PART_LEN2],
float* echo_subtractor_output,
@ -1279,6 +1268,13 @@ static void ProcessBlock(AecCore* aec) {
}
#endif
if (aec->metricsMode == 1) {
// Update power levels
UpdateLevel(&aec->farlevel,
CalculatePower(&farend_ptr[PART_LEN], PART_LEN));
UpdateLevel(&aec->nearlevel, CalculatePower(nearend_ptr, PART_LEN));
}
// Convert far-end signal to the frequency domain.
memcpy(fft, farend_ptr, sizeof(float) * PART_LEN2);
Fft(fft, xf);
@ -1288,12 +1284,6 @@ static void ProcessBlock(AecCore* aec) {
memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
Fft(fft, df);
if (aec->metricsMode == 1) {
// Update power levels
UpdateLevel(&aec->farlevel, CalculatePower(farend_ptr, PART_LEN2));
UpdateLevel(&aec->nearlevel, CalculatePower(aec->dBuf, PART_LEN2));
}
// Power smoothing
for (i = 0; i < PART_LEN1; i++) {
far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
@ -1374,7 +1364,6 @@ static void ProcessBlock(AecCore* aec) {
EchoSubtraction(aec,
aec->num_partitions,
aec->xfBufBlockPos,
aec->metricsMode,
aec->extended_filter_enabled,
aec->normal_mu,
aec->normal_error_threshold,
@ -1382,11 +1371,15 @@ static void ProcessBlock(AecCore* aec) {
nearend_ptr,
aec->xPow,
aec->wfBuf,
&aec->linoutlevel,
echo_subtractor_output);
RTC_AEC_DEBUG_WAV_WRITE(aec->outLinearFile, echo_subtractor_output, PART_LEN);
if (aec->metricsMode == 1) {
UpdateLevel(&aec->linoutlevel,
CalculatePower(echo_subtractor_output, PART_LEN));
}
// Perform echo suppression.
EchoSuppression(aec, farend_ptr, echo_subtractor_output, output, outputH_ptr);
@ -1713,7 +1706,6 @@ int WebRtcAec_InitAec(AecCore* aec, int sampFreq) {
return 0;
}
// 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.

4
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
View File

@ -203,10 +203,10 @@ int16_t MaxAudioFrame(const AudioFrame& frame) {
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
void TestStats(const AudioProcessing::Statistic& test,
const audioproc::Test::Statistic& reference) {
EXPECT_EQ(reference.instant(), test.instant);
EXPECT_NEAR(reference.instant(), test.instant, 1);
EXPECT_EQ(reference.average(), test.average);
EXPECT_EQ(reference.maximum(), test.maximum);
EXPECT_EQ(reference.minimum(), test.minimum);
EXPECT_NEAR(reference.minimum(), test.minimum, 1);
}
void WriteStatsMessage(const AudioProcessing::Statistic& output,