Revert 4837 "Add an extended filter mode to AEC."
> Add an extended filter mode to AEC.
>
> This mode extends the filter length from the current 48 ms to 128 ms.
> It is runtime selectable which allows it to be enabled through
> experiment. We reuse the DelayCorrection infrastructure to avoid having
> to replumb everything up to libjingle.
>
> Increases AEC complexity by ~50% on modern x86 CPUs.
> Measurements (in percent of usage on one core):
>
> Machine/CPU Normal Extended
> MacBook Retina (Early 2013),
> Core i7 Ivy Bridge (2.7 GHz, hyperthreaded) 0.6% 0.9%
>
> MacBook Air (Late 2010), Core 2 Duo (2.13 GHz) 1.4% 2.7%
>
> Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz) 0.6% 1.0%
>
> Samsung ARM Chromebook,
> Samsung Exynos 5 Dual (1.7 GHz) 3.2% 5.6%
>
> The relative value is large of course but the absolute should be
> acceptable in order to have a working AEC on some platforms.
>
> Detailed changes to the algorithm:
> - The filter length is changed from 48 to 128 ms. This comes with tuning
> of several parameters: i) filter adaptation stepsize and error
> threshold; ii) non-linear processing smoothing and overdrive.
> - Option to ignore the reported delays on platforms which we deem
> sufficiently unreliable. Currently this will be enabled in Chromium for
> Mac.
> - Faster startup times by removing the excessive "startup phase"
> processing of reported delays.
> - Much more conservative adjustments to the far-end read pointer. We
> smooth the delay difference more heavily, and back off from the
> difference more. Adjustments force a readaptation of the filter, so they
> should be avoided except when really necessary.
>
> Corresponds to these changes:
> https://chromereviews.googleplex.com/9412014
> https://chromereviews.googleplex.com/9514013
> https://chromereviews.googleplex.com/9960013
>
> BUG=454,827,1261
> R=bjornv@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/2151007
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4839 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 40e9f67..d194c82 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -109,17 +109,7 @@
// Target suppression levels for nlp modes.
// log{0.001, 0.00001, 0.00000001}
static const float kTargetSupp[3] = { -6.9f, -11.5f, -18.4f };
-
-// Two sets of parameters, one for the extended filter mode.
-static const float kExtendedMinOverDrive[3] = { 3.0f, 6.0f, 15.0f };
-static const float kNormalMinOverDrive[3] = { 1.0f, 2.0f, 5.0f };
-static const float kExtendedSmoothingCoefficients[2][2] =
- { { 0.9f, 0.1f }, { 0.92f, 0.08f } };
-static const float kNormalSmoothingCoefficients[2][2] =
- { { 0.9f, 0.1f }, { 0.93f, 0.07f } };
-
-// Number of partitions forming the NLP's "preferred" bands.
-enum { kPrefBandSize = 24 };
+static const float kMinOverDrive[3] = { 1.0f, 2.0f, 5.0f };
#ifdef WEBRTC_AEC_DEBUG_DUMP
extern int webrtc_aec_instance_count;
@@ -291,13 +281,13 @@
static void FilterFar(AecCore* aec, float yf[2][PART_LEN1])
{
int i;
- for (i = 0; i < aec->num_partitions; i++) {
+ for (i = 0; i < NR_PART; i++) {
int j;
int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
int pos = i * PART_LEN1;
// Check for wrap
- if (i + aec->xfBufBlockPos >= aec->num_partitions) {
- xPos -= aec->num_partitions*(PART_LEN1);
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART*(PART_LEN1);
}
for (j = 0; j < PART_LEN1; j++) {
@@ -311,25 +301,22 @@
static void ScaleErrorSignal(AecCore* aec, float ef[2][PART_LEN1])
{
- const float mu = aec->extended_filter_enabled ? kExtendedMu : aec->normal_mu;
- const float error_threshold = aec->extended_filter_enabled ?
- kExtendedErrorThreshold : aec->normal_error_threshold;
int i;
- float abs_ef;
+ float absEf;
for (i = 0; i < (PART_LEN1); i++) {
ef[0][i] /= (aec->xPow[i] + 1e-10f);
ef[1][i] /= (aec->xPow[i] + 1e-10f);
- abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+ absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
- if (abs_ef > error_threshold) {
- abs_ef = error_threshold / (abs_ef + 1e-10f);
- ef[0][i] *= abs_ef;
- ef[1][i] *= abs_ef;
+ if (absEf > aec->errThresh) {
+ absEf = aec->errThresh / (absEf + 1e-10f);
+ ef[0][i] *= absEf;
+ ef[1][i] *= absEf;
}
// Stepsize factor
- ef[0][i] *= mu;
- ef[1][i] *= mu;
+ ef[0][i] *= aec->mu;
+ ef[1][i] *= aec->mu;
}
}
@@ -338,35 +325,35 @@
//static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
// float ef[2][PART_LEN1]) {
// int i, j;
-// for (i = 0; i < aec->num_partitions; i++) {
+// for (i = 0; i < NR_PART; i++) {
// int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
// int pos;
// // Check for wrap
-// if (i + aec->xfBufBlockPos >= aec->num_partitions) {
-// xPos -= aec->num_partitions * PART_LEN1;
+// if (i + aec->xfBufBlockPos >= NR_PART) {
+// xPos -= NR_PART * PART_LEN1;
// }
//
// pos = i * PART_LEN1;
//
// for (j = 0; j < PART_LEN1; j++) {
-// aec->wfBuf[0][pos + j] += MulRe(aec->xfBuf[0][xPos + j],
-// -aec->xfBuf[1][xPos + j],
-// ef[0][j], ef[1][j]);
-// aec->wfBuf[1][pos + j] += MulIm(aec->xfBuf[0][xPos + j],
-// -aec->xfBuf[1][xPos + j],
-// ef[0][j], ef[1][j]);
+// aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
+// aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
+// -aec->xfBuf[xPos + j][1],
+// ef[j][0], ef[j][1]);
// }
// }
//}
static void FilterAdaptation(AecCore* aec, float *fft, float ef[2][PART_LEN1]) {
int i, j;
- for (i = 0; i < aec->num_partitions; i++) {
+ for (i = 0; i < NR_PART; i++) {
int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
int pos;
// Check for wrap
- if (i + aec->xfBufBlockPos >= aec->num_partitions) {
- xPos -= aec->num_partitions * PART_LEN1;
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART * PART_LEN1;
}
pos = i * PART_LEN1;
@@ -440,12 +427,12 @@
aec->sampFreq = sampFreq;
if (sampFreq == 8000) {
- aec->normal_mu = 0.6f;
- aec->normal_error_threshold = 2e-6f;
+ aec->mu = 0.6f;
+ aec->errThresh = 2e-6f;
}
else {
- aec->normal_mu = 0.5f;
- aec->normal_error_threshold = 1.5e-6f;
+ aec->mu = 0.5f;
+ aec->errThresh = 1.5e-6f;
}
if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
@@ -487,9 +474,6 @@
aec->delay_logging_enabled = 0;
memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
- aec->extended_filter_enabled = 0;
- aec->num_partitions = kNormalNumPartitions;
-
// Default target suppression mode.
aec->nlp_mode = 1;
@@ -499,7 +483,7 @@
aec->mult = (short)aec->sampFreq / 16000;
}
else {
- aec->mult = (short)aec->sampFreq / 8000;
+ aec->mult = (short)aec->sampFreq / 8000;
}
aec->farBufWritePos = 0;
@@ -530,14 +514,11 @@
aec->xfBufBlockPos = 0;
// TODO: Investigate need for these initializations. Deleting them doesn't
// change the output at all and yields 0.4% overall speedup.
- memset(aec->xfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
- PART_LEN1);
- memset(aec->wfBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
- PART_LEN1);
+ memset(aec->xfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
+ memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
- memset(aec->xfwBuf, 0, sizeof(complex_t) * kExtendedNumPartitions *
- PART_LEN1);
+ memset(aec->xfwBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
memset(aec->se, 0, sizeof(float) * PART_LEN1);
// To prevent numerical instability in the first block.
@@ -753,11 +734,13 @@
}
int WebRtcAec_echo_state(AecCore* self) {
+ assert(self != NULL);
return self->echoState;
}
void WebRtcAec_GetEchoStats(AecCore* self, Stats* erl, Stats* erle,
Stats* a_nlp) {
+ assert(self != NULL);
assert(erl != NULL);
assert(erle != NULL);
assert(a_nlp != NULL);
@@ -768,12 +751,14 @@
#ifdef WEBRTC_AEC_DEBUG_DUMP
void* WebRtcAec_far_time_buf(AecCore* self) {
+ assert(self != NULL);
return self->far_time_buf;
}
#endif
void WebRtcAec_SetConfigCore(AecCore* self, int nlp_mode, int metrics_mode,
int delay_logging) {
+ assert(self != NULL);
assert(nlp_mode >= 0 && nlp_mode < 3);
self->nlp_mode = nlp_mode;
self->metricsMode = metrics_mode;
@@ -786,20 +771,13 @@
}
}
-void WebRtcAec_enable_delay_correction(AecCore* self, int enable) {
- self->extended_filter_enabled = enable;
- self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
-}
-
-int WebRtcAec_delay_correction_enabled(AecCore* self) {
- return self->extended_filter_enabled;
-}
-
int WebRtcAec_system_delay(AecCore* self) {
+ assert(self != NULL);
return self->system_delay;
}
void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
+ assert(self != NULL);
assert(delay >= 0);
self->system_delay = delay;
}
@@ -875,8 +853,7 @@
for (i = 0; i < PART_LEN1; i++) {
far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
(xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
- aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * aec->num_partitions *
- far_spectrum;
+ aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART * far_spectrum;
// Calculate absolute spectra
abs_far_spectrum[i] = sqrtf(far_spectrum);
@@ -936,7 +913,7 @@
// Update the xfBuf block position.
aec->xfBufBlockPos--;
if (aec->xfBufBlockPos == -1) {
- aec->xfBufBlockPos = aec->num_partitions - 1;
+ aec->xfBufBlockPos = NR_PART - 1;
}
// Buffer xf
@@ -1037,21 +1014,18 @@
float cohde[PART_LEN1], cohxd[PART_LEN1];
float hNlDeAvg, hNlXdAvg;
float hNl[PART_LEN1];
- float hNlPref[kPrefBandSize];
+ float hNlPref[PREF_BAND_SIZE];
float hNlFb = 0, hNlFbLow = 0;
const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
- const int prefBandSize = kPrefBandSize / aec->mult;
+ const int prefBandSize = PREF_BAND_SIZE / aec->mult;
const int minPrefBand = 4 / aec->mult;
// Near and error power sums
float sdSum = 0, seSum = 0;
- // Power estimate smoothing coefficients.
- const float *ptrGCoh = aec->extended_filter_enabled ?
- kExtendedSmoothingCoefficients[aec->mult - 1] :
- kNormalSmoothingCoefficients[aec->mult - 1];
- const float* min_overdrive = aec->extended_filter_enabled ?
- kExtendedMinOverDrive : kNormalMinOverDrive;
+ // Power estimate smoothing coefficients
+ const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
+ const float *ptrGCoh = gCoh[aec->mult - 1];
// Filter energy
float wfEnMax = 0, wfEn = 0;
@@ -1074,7 +1048,7 @@
if (aec->delayEstCtr == 0) {
wfEnMax = 0;
aec->delayIdx = 0;
- for (i = 0; i < aec->num_partitions; i++) {
+ for (i = 0; i < NR_PART; i++) {
pos = i * PART_LEN1;
wfEn = 0;
for (j = 0; j < PART_LEN1; j++) {
@@ -1215,7 +1189,7 @@
if (aec->hNlXdAvgMin == 1) {
aec->echoState = 0;
- aec->overDrive = min_overdrive[aec->nlp_mode];
+ aec->overDrive = kMinOverDrive[aec->nlp_mode];
if (aec->stNearState == 1) {
memcpy(hNl, cohde, sizeof(hNl));
@@ -1271,7 +1245,7 @@
aec->hNlMinCtr = 0;
aec->overDrive = WEBRTC_SPL_MAX(kTargetSupp[aec->nlp_mode] /
((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f),
- min_overdrive[aec->nlp_mode]);
+ kMinOverDrive[aec->nlp_mode]);
}
// Smooth the overdrive.
@@ -1491,6 +1465,7 @@
}
static void InitMetrics(AecCore* self) {
+ assert(self != NULL);
self->stateCounter = 0;
InitLevel(&self->farlevel);
InitLevel(&self->nearlevel);
@@ -1712,4 +1687,3 @@
freq_data[1][i] = time_data[2 * i + 1];
}
}
-
diff --git a/webrtc/modules/audio_processing/aec/aec_core.h b/webrtc/modules/audio_processing/aec/aec_core.h
index f83c37c..6380717 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/webrtc/modules/audio_processing/aec/aec_core.h
@@ -70,38 +70,23 @@
// Returns the number of elements moved, and adjusts |system_delay| by the
// corresponding amount in ms.
int WebRtcAec_MoveFarReadPtr(AecCore* aec, int elements);
-
// Calculates the median and standard deviation among the delay estimates
// collected since the last call to this function.
int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std);
-
// Returns the echo state (1: echo, 0: no echo).
int WebRtcAec_echo_state(AecCore* self);
-
// Gets statistics of the echo metrics ERL, ERLE, A_NLP.
void WebRtcAec_GetEchoStats(AecCore* self, Stats* erl, Stats* erle,
Stats* a_nlp);
#ifdef WEBRTC_AEC_DEBUG_DUMP
void* WebRtcAec_far_time_buf(AecCore* self);
#endif
-
// Sets local configuration modes.
void WebRtcAec_SetConfigCore(AecCore* self, int nlp_mode, int metrics_mode,
int delay_logging);
-
-// We now interpret delay correction to mean an extended filter length feature.
-// We reuse the delay correction infrastructure to avoid changes through to
-// libjingle. See details along with |DelayCorrection| in
-// echo_cancellation_impl.h. Non-zero enables, zero disables.
-void WebRtcAec_enable_delay_correction(AecCore* self, int enable);
-
-// Returns non-zero if delay correction is enabled and zero if disabled.
-int WebRtcAec_delay_correction_enabled(AecCore* self);
-
// Returns the current |system_delay|, i.e., the buffered difference between
// far-end and near-end.
int WebRtcAec_system_delay(AecCore* self);
-
// Sets the |system_delay| to |value|. Note that if the value is changed
// improperly, there can be a performance regression. So it should be used with
// care.
diff --git a/webrtc/modules/audio_processing/aec/aec_core_internal.h b/webrtc/modules/audio_processing/aec/aec_core_internal.h
index 4480101..3b92bd6 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_internal.h
+++ b/webrtc/modules/audio_processing/aec/aec_core_internal.h
@@ -19,15 +19,8 @@
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include "webrtc/typedefs.h"
-// Number of partitions for the extended filter mode. The first one is an enum
-// to be used in array declarations, as it represents the maximum filter length.
-enum { kExtendedNumPartitions = 32 };
-static const int kNormalNumPartitions = 12;
-
-// Extended filter adaptation parameters.
-// TODO(ajm): No narrowband tuning yet.
-static const float kExtendedMu = 0.4f;
-static const float kExtendedErrorThreshold = 1.0e-6f;
+#define NR_PART 12 // Number of partitions in filter.
+#define PREF_BAND_SIZE 24
typedef struct PowerLevel {
float sfrsum;
@@ -63,12 +56,11 @@
float dInitMinPow[PART_LEN1];
float *noisePow;
- float xfBuf[2][kExtendedNumPartitions * PART_LEN1]; // farend fft buffer
- float wfBuf[2][kExtendedNumPartitions * PART_LEN1]; // filter fft
+ float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
+ float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
complex_t sde[PART_LEN1]; // cross-psd of nearend and error
complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
- // Farend windowed fft buffer.
- complex_t xfwBuf[kExtendedNumPartitions * PART_LEN1];
+ complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near, error psd
float hNs[PART_LEN1];
@@ -93,8 +85,8 @@
int sampFreq;
uint32_t seed;
- float normal_mu; // stepsize
- float normal_error_threshold; // error threshold
+ float mu; // stepsize
+ float errThresh; // error threshold
int noiseEstCtr;
@@ -120,11 +112,6 @@
void* delay_estimator_farend;
void* delay_estimator;
- // 1 = extended filter mode enabled, 0 = disabled.
- int extended_filter_enabled;
- // Runtime selection of number of filter partitions.
- int num_partitions;
-
#ifdef WEBRTC_AEC_DEBUG_DUMP
RingBuffer* far_time_buf;
FILE *farFile;
diff --git a/webrtc/modules/audio_processing/aec/aec_core_sse2.c b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
index 61602a8..fdc6872 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_sse2.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
@@ -34,14 +34,13 @@
static void FilterFarSSE2(AecCore* aec, float yf[2][PART_LEN1])
{
int i;
- const int num_partitions = aec->num_partitions;
- for (i = 0; i < num_partitions; i++) {
+ for (i = 0; i < NR_PART; i++) {
int j;
int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
int pos = i * PART_LEN1;
// Check for wrap
- if (i + aec->xfBufBlockPos >= num_partitions) {
- xPos -= num_partitions*(PART_LEN1);
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART*(PART_LEN1);
}
// vectorized code (four at once)
@@ -76,11 +75,8 @@
static void ScaleErrorSignalSSE2(AecCore* aec, float ef[2][PART_LEN1])
{
const __m128 k1e_10f = _mm_set1_ps(1e-10f);
- const __m128 kMu = aec->extended_filter_enabled ?
- _mm_set1_ps(kExtendedMu) : _mm_set1_ps(aec->normal_mu);
- const __m128 kThresh = aec->extended_filter_enabled ?
- _mm_set1_ps(kExtendedErrorThreshold) :
- _mm_set1_ps(aec->normal_error_threshold);
+ const __m128 kThresh = _mm_set1_ps(aec->errThresh);
+ const __m128 kMu = _mm_set1_ps(aec->mu);
int i;
// vectorized code (four at once)
@@ -114,39 +110,32 @@
_mm_storeu_ps(&ef[1][i], ef_im);
}
// scalar code for the remaining items.
- {
- const float mu = aec->extended_filter_enabled ?
- kExtendedMu : aec->normal_mu;
- const float error_threshold = aec->extended_filter_enabled ?
- kExtendedErrorThreshold : aec->normal_error_threshold;
- for (; i < (PART_LEN1); i++) {
- float abs_ef;
- ef[0][i] /= (aec->xPow[i] + 1e-10f);
- ef[1][i] /= (aec->xPow[i] + 1e-10f);
- abs_ef = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
+ for (; i < (PART_LEN1); i++) {
+ float absEf;
+ ef[0][i] /= (aec->xPow[i] + 1e-10f);
+ ef[1][i] /= (aec->xPow[i] + 1e-10f);
+ absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
- if (abs_ef > error_threshold) {
- abs_ef = error_threshold / (abs_ef + 1e-10f);
- ef[0][i] *= abs_ef;
- ef[1][i] *= abs_ef;
- }
-
- // Stepsize factor
- ef[0][i] *= mu;
- ef[1][i] *= mu;
+ if (absEf > aec->errThresh) {
+ absEf = aec->errThresh / (absEf + 1e-10f);
+ ef[0][i] *= absEf;
+ ef[1][i] *= absEf;
}
+
+ // Stepsize factor
+ ef[0][i] *= aec->mu;
+ ef[1][i] *= aec->mu;
}
}
static void FilterAdaptationSSE2(AecCore* aec, float *fft, float ef[2][PART_LEN1]) {
int i, j;
- const int num_partitions = aec->num_partitions;
- for (i = 0; i < num_partitions; i++) {
+ for (i = 0; i < NR_PART; i++) {
int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
int pos = i * PART_LEN1;
// Check for wrap
- if (i + aec->xfBufBlockPos >= num_partitions) {
- xPos -= num_partitions * PART_LEN1;
+ if (i + aec->xfBufBlockPos >= NR_PART) {
+ xPos -= NR_PART * PART_LEN1;
}
// Process the whole array...
@@ -424,4 +413,3 @@
WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
}
-
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 07bca55..2d41359 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -27,61 +27,6 @@
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include "webrtc/typedefs.h"
-// Measured delays [ms]
-// Device Chrome GTP
-// MacBook Air 10
-// MacBook Retina 10 100
-// MacPro 30?
-//
-// Win7 Desktop 70 80?
-// Win7 T430s 110
-// Win8 T420s 70
-//
-// Daisy 50
-// Pixel (w/ preproc?) 240
-// Pixel (w/o preproc?) 110 110
-
-// The extended filter mode gives us the flexibility to ignore the system's
-// reported delays. We do this for platforms which we believe provide results
-// which are incompatible with the AEC's expectations. Based on measurements
-// (some provided above) we set a conservative (i.e. lower than measured)
-// fixed delay.
-//
-// WEBRTC_UNTRUSTED_DELAY will only have an impact when |extended_filter_mode|
-// is enabled. See the note along with |DelayCorrection| in
-// echo_cancellation_impl.h for more details on the mode.
-//
-// Justification:
-// Chromium/Mac: Here, the true latency is so low (~10-20 ms), that it plays
-// havoc with the AEC's buffering. To avoid this, we set a fixed delay of 20 ms
-// and then compensate by rewinding by 10 ms (in wideband) through
-// kDelayDiffOffsetSamples. This trick does not seem to work for larger rewind
-// values, but fortunately this is sufficient.
-//
-// Chromium/Linux(ChromeOS): The values we get on this platform don't correspond
-// well to reality. The variance doesn't match the AEC's buffer changes, and the
-// bulk values tend to be too low. However, the range across different hardware
-// appears to be too large to choose a single value.
-//
-// GTP/Linux(ChromeOS): TBD, but for the moment we will trust the values.
-#if defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_MAC)
-#define WEBRTC_UNTRUSTED_DELAY
-#endif
-
-#if defined(WEBRTC_MAC)
-static const int kFixedDelayMs = 20;
-static const int kDelayDiffOffsetSamples = -160;
-#elif defined(WEBRTC_WIN)
-static const int kFixedDelayMs = 50;
-static const int kDelayDiffOffsetSamples = 0;
-#else
-// Essentially ChromeOS.
-static const int kFixedDelayMs = 50;
-static const int kDelayDiffOffsetSamples = 0;
-#endif
-static const int kMinTrustedDelayMs = 20;
-static const int kMaxTrustedDelayMs = 500;
-
// Maximum length of resampled signal. Must be an integer multiple of frames
// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
// The factor of 2 handles wb, and the + 1 is as a safety margin
@@ -98,14 +43,7 @@
// Estimates delay to set the position of the far-end buffer read pointer
// (controlled by knownDelay)
-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,
- 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,
- int16_t num_samples, int16_t reported_delay_ms, int32_t skew);
+static int EstBufDelay(aecpc_t *aecInst);
int32_t WebRtcAec_Create(void **aecInst)
{
@@ -197,6 +135,10 @@
aecpc_t *aecpc = aecInst;
AecConfig aecConfig;
+ if (aecpc == NULL) {
+ return -1;
+ }
+
if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
return -1;
@@ -235,31 +177,31 @@
aecpc->splitSampFreq = sampFreq;
}
+ aecpc->skewFrCtr = 0;
+ aecpc->activity = 0;
+
aecpc->delayCtr = 0;
- aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
- // Sampling frequency multiplier (SWB is processed as 160 frame size).
- aecpc->rate_factor = aecpc->splitSampFreq / 8000;
aecpc->sum = 0;
aecpc->counter = 0;
aecpc->checkBuffSize = 1;
aecpc->firstVal = 0;
- aecpc->startup_phase = 1;
+ aecpc->ECstartup = 1;
aecpc->bufSizeStart = 0;
aecpc->checkBufSizeCtr = 0;
- aecpc->msInSndCardBuf = 0;
- aecpc->filtDelay = -1; // -1 indicates an initialized state.
+ aecpc->filtDelay = 0;
aecpc->timeForDelayChange = 0;
aecpc->knownDelay = 0;
aecpc->lastDelayDiff = 0;
- aecpc->skewFrCtr = 0;
+ aecpc->skew = 0;
aecpc->resample = kAecFalse;
aecpc->highSkewCtr = 0;
- aecpc->skew = 0;
+ aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
- aecpc->farend_started = 0;
+ // Sampling frequency multiplier (SWB is processed as 160 frame size).
+ aecpc->rate_factor = aecpc->splitSampFreq / 8000;
// Default settings.
aecConfig.nlpMode = kAecNlpModerate;
@@ -297,6 +239,10 @@
float skew;
int i = 0;
+ if (aecpc == NULL) {
+ return -1;
+ }
+
if (farend == NULL) {
aecpc->lastError = AEC_NULL_POINTER_ERROR;
return -1;
@@ -322,7 +268,6 @@
farend_ptr = (const int16_t*) newFarend;
}
- aecpc->farend_started = 1;
WebRtcAec_SetSystemDelay(aecpc->aec, WebRtcAec_system_delay(aecpc->aec) +
newNrOfSamples);
@@ -366,6 +311,17 @@
{
aecpc_t *aecpc = aecInst;
int32_t 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;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
if (nearend == NULL) {
aecpc->lastError = AEC_NULL_POINTER_ERROR;
return -1;
@@ -398,21 +354,144 @@
aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
retVal = -1;
}
- else if (msInSndCardBuf > kMaxTrustedDelayMs) {
- // The clamping is now done in ProcessExtended/Normal().
+ else if (msInSndCardBuf > 500) {
+ msInSndCardBuf = 500;
aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
retVal = -1;
}
+ // TODO(andrew): we need to investigate if this +10 is really wanted.
+ msInSndCardBuf += 10;
+ aecpc->msInSndCardBuf = msInSndCardBuf;
- // 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);
+ if (aecpc->skewMode == kAecTrue) {
+ if (aecpc->skewFrCtr < 25) {
+ aecpc->skewFrCtr++;
+ }
+ else {
+ retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
+ if (retVal == -1) {
+ aecpc->skew = 0;
+ aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
+ }
+
+ aecpc->skew /= aecpc->sampFactor*nrOfSamples;
+
+ if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
+ aecpc->resample = kAecFalse;
+ }
+ else {
+ aecpc->resample = kAecTrue;
+ }
+
+ if (aecpc->skew < minSkewEst) {
+ aecpc->skew = minSkewEst;
+ }
+ else if (aecpc->skew > maxSkewEst) {
+ aecpc->skew = maxSkewEst;
+ }
+
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+ (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
+#endif
+ }
+ }
+
+ nFrames = nrOfSamples / FRAME_LEN;
+ nBlocks10ms = nFrames / aecpc->rate_factor;
+
+ if (aecpc->ECstartup) {
+ if (nearend != out) {
+ // Only needed if they don't already point to the same place.
+ memcpy(out, nearend, sizeof(short) * nrOfSamples);
+ }
+
+ // The AEC is in the start up mode
+ // AEC is disabled until the system delay is OK
+
+ // Mechanism to ensure that the system delay is reasonably stable.
+ if (aecpc->checkBuffSize) {
+ aecpc->checkBufSizeCtr++;
+ // Before we fill up the far-end buffer we require the system delay
+ // to be stable (+/-8 ms) compared to the first value. This
+ // comparison is made during the following 6 consecutive 10 ms
+ // blocks. If it seems to be stable then we start to fill up the
+ // far-end buffer.
+ if (aecpc->counter == 0) {
+ aecpc->firstVal = aecpc->msInSndCardBuf;
+ aecpc->sum = 0;
+ }
+
+ if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
+ WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
+ aecpc->sum += aecpc->msInSndCardBuf;
+ aecpc->counter++;
+ }
+ else {
+ aecpc->counter = 0;
+ }
+
+ if (aecpc->counter * nBlocks10ms >= 6) {
+ // The far-end buffer size is determined in partitions of
+ // PART_LEN samples. Use 75% of the average value of the system
+ // delay as buffer size to start with.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
+ aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN),
+ kMaxBufSizeStart);
+ // Buffer size has now been determined.
+ aecpc->checkBuffSize = 0;
+ }
+
+ if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
+ // For really bad systems, don't disable the echo canceller for
+ // more than 0.5 sec.
+ aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
+ aecpc->rate_factor * 3) / 40, kMaxBufSizeStart);
+ aecpc->checkBuffSize = 0;
+ }
+ }
+
+ // If |checkBuffSize| changed in the if-statement above.
+ if (!aecpc->checkBuffSize) {
+ // The system delay is now reasonably stable (or has been unstable
+ // for too long). When the far-end buffer is filled with
+ // approximately the same amount of data as reported by the system
+ // we end the startup phase.
+ int overhead_elements =
+ WebRtcAec_system_delay(aecpc->aec) / PART_LEN -
+ aecpc->bufSizeStart;
+ if (overhead_elements == 0) {
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ } else if (overhead_elements > 0) {
+ // TODO(bjornv): Do we need a check on how much we actually
+ // moved the read pointer? It should always be possible to move
+ // the pointer |overhead_elements| since we have only added data
+ // to the buffer and no delay compensation nor AEC processing
+ // has been done.
+ WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements);
+
+ // Enable the AEC
+ aecpc->ECstartup = 0;
+ }
+ }
} else {
- if (ProcessNormal(aecpc, nearend, nearendH, out, outH, nrOfSamples,
- msInSndCardBuf, skew) != 0) {
- retVal = -1;
- }
+ // AEC is enabled.
+
+ EstBufDelay(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|.
+ }
}
#ifdef WEBRTC_AEC_DEBUG_DUMP
@@ -430,6 +509,11 @@
int WebRtcAec_set_config(void* handle, AecConfig config) {
aecpc_t* self = (aecpc_t*)handle;
+
+ if (handle == NULL ) {
+ return -1;
+ }
+
if (self->initFlag != initCheck) {
self->lastError = AEC_UNINITIALIZED_ERROR;
return -1;
@@ -464,6 +548,10 @@
int WebRtcAec_get_echo_status(void* handle, int* status) {
aecpc_t* self = (aecpc_t*)handle;
+
+ if (handle == NULL ) {
+ return -1;
+ }
if (status == NULL ) {
self->lastError = AEC_NULL_POINTER_ERROR;
return -1;
@@ -577,6 +665,10 @@
int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
aecpc_t* self = handle;
+
+ if (handle == NULL) {
+ return -1;
+ }
if (median == NULL) {
self->lastError = AEC_NULL_POINTER_ERROR;
return -1;
@@ -601,6 +693,11 @@
int32_t WebRtcAec_get_error_code(void *aecInst)
{
aecpc_t *aecpc = aecInst;
+
+ if (aecpc == NULL) {
+ return -1;
+ }
+
return aecpc->lastError;
}
@@ -611,220 +708,7 @@
return ((aecpc_t*) handle)->aec;
}
-static int ProcessNormal(aecpc_t *aecpc, const int16_t *nearend,
- const int16_t *nearendH, int16_t *out, int16_t *outH,
- 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;
-
- msInSndCardBuf = msInSndCardBuf > kMaxTrustedDelayMs ?
- kMaxTrustedDelayMs : msInSndCardBuf;
- // TODO(andrew): we need to investigate if this +10 is really wanted.
- msInSndCardBuf += 10;
- aecpc->msInSndCardBuf = msInSndCardBuf;
-
- if (aecpc->skewMode == kAecTrue) {
- if (aecpc->skewFrCtr < 25) {
- aecpc->skewFrCtr++;
- }
- else {
- retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
- if (retVal == -1) {
- aecpc->skew = 0;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- }
-
- aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
- if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
- aecpc->resample = kAecFalse;
- }
- else {
- aecpc->resample = kAecTrue;
- }
-
- if (aecpc->skew < minSkewEst) {
- aecpc->skew = minSkewEst;
- }
- else if (aecpc->skew > maxSkewEst) {
- aecpc->skew = maxSkewEst;
- }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- (void)fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
- }
- }
-
- nFrames = nrOfSamples / FRAME_LEN;
- nBlocks10ms = nFrames / 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(short) * nrOfSamples);
- }
- if (nearendH != outH) {
- memcpy(outH, nearendH, sizeof(short) * nrOfSamples);
- }
-
- // The AEC is in the start up mode
- // AEC is disabled until the system delay is OK
-
- // Mechanism to ensure that the system delay is reasonably stable.
- if (aecpc->checkBuffSize) {
- aecpc->checkBufSizeCtr++;
- // Before we fill up the far-end buffer we require the system delay
- // to be stable (+/-8 ms) compared to the first value. This
- // comparison is made during the following 6 consecutive 10 ms
- // blocks. If it seems to be stable then we start to fill up the
- // far-end buffer.
- if (aecpc->counter == 0) {
- aecpc->firstVal = aecpc->msInSndCardBuf;
- aecpc->sum = 0;
- }
-
- if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
- WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
- aecpc->sum += aecpc->msInSndCardBuf;
- aecpc->counter++;
- }
- else {
- aecpc->counter = 0;
- }
-
- if (aecpc->counter * nBlocks10ms >= 6) {
- // The far-end buffer size is determined in partitions of
- // PART_LEN samples. Use 75% of the average value of the system
- // delay as buffer size to start with.
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
- aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN),
- kMaxBufSizeStart);
- // Buffer size has now been determined.
- aecpc->checkBuffSize = 0;
- }
-
- if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
- // For really bad systems, don't disable the echo canceller for
- // more than 0.5 sec.
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
- aecpc->rate_factor * 3) / 40, kMaxBufSizeStart);
- aecpc->checkBuffSize = 0;
- }
- }
-
- // If |checkBuffSize| changed in the if-statement above.
- if (!aecpc->checkBuffSize) {
- // The system delay is now reasonably stable (or has been unstable
- // for too long). When the far-end buffer is filled with
- // approximately the same amount of data as reported by the system
- // we end the startup phase.
- int overhead_elements =
- WebRtcAec_system_delay(aecpc->aec) / PART_LEN - aecpc->bufSizeStart;
- if (overhead_elements == 0) {
- // Enable the AEC
- aecpc->startup_phase = 0;
- } else if (overhead_elements > 0) {
- // TODO(bjornv): Do we need a check on how much we actually
- // moved the read pointer? It should always be possible to move
- // the pointer |overhead_elements| since we have only added data
- // to the buffer and no delay compensation nor AEC processing
- // has been done.
- WebRtcAec_MoveFarReadPtr(aecpc->aec, overhead_elements);
-
- // Enable the AEC
- aecpc->startup_phase = 0;
- }
- }
- } else {
- // AEC is enabled.
- 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|.
- }
- }
-
- return retVal;
-}
-
-static void ProcessExtended(aecpc_t* self, const int16_t* near,
- const int16_t* near_high, int16_t* out, int16_t* out_high,
- int16_t num_samples, int16_t reported_delay_ms, int32_t skew) {
- int i;
- const int num_frames = num_samples / FRAME_LEN;
-#if defined(WEBRTC_UNTRUSTED_DELAY)
- const int delay_diff_offset = kDelayDiffOffsetSamples;
- reported_delay_ms = kFixedDelayMs;
-#else
- // This is the usual mode where we trust the reported system delay values.
- const int delay_diff_offset = 0;
- // Due to the longer filter, we no longer add 10 ms to the reported delay
- // to reduce chance of non-causality. Instead we apply a minimum here to avoid
- // issues with the read pointer jumping around needlessly.
- reported_delay_ms = reported_delay_ms < kMinTrustedDelayMs ?
- kMinTrustedDelayMs : reported_delay_ms;
- // If the reported delay appears to be bogus, we attempt to recover by using
- // the measured fixed delay values. We use >= here because higher layers
- // may already clamp to this maximum value, and we would otherwise not
- // detect it here.
- reported_delay_ms = reported_delay_ms >= kMaxTrustedDelayMs ?
- kFixedDelayMs : reported_delay_ms;
-#endif
- 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(short) * num_samples);
- }
- if (near_high != out_high) {
- memcpy(out_high, near_high, sizeof(short) * num_samples);
- }
- return;
- }
- if (self->startup_phase) {
- // In the extended mode, there isn't a startup "phase", just a special
- // action on the first frame. In the trusted delay case, we'll take the
- // current reported delay, unless it's less then our conservative
- // measurement.
- int startup_size_ms = reported_delay_ms < kFixedDelayMs ?
- kFixedDelayMs : reported_delay_ms;
- int overhead_elements = (WebRtcAec_system_delay(self->aec) -
- startup_size_ms / 2 * self->rate_factor * 8) / PART_LEN;
- WebRtcAec_MoveFarReadPtr(self->aec, overhead_elements);
- self->startup_phase = 0;
- }
-
- EstBufDelayExtended(self);
-
- for (i = 0; i < num_frames; ++i) {
- // |delay_diff_offset| gives us the option to manually rewind the delay on
- // very low delay platforms which can't be expressed purely through
- // |reported_delay_ms|.
- WebRtcAec_ProcessFrame(self->aec, &near[FRAME_LEN * i],
- &near_high[FRAME_LEN * i], self->knownDelay + delay_diff_offset,
- &out[FRAME_LEN * i], &out_high[FRAME_LEN * i]);
- }
-}
-
-static void EstBufDelayNormal(aecpc_t* aecpc) {
+static int EstBufDelay(aecpc_t* aecpc) {
int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->rate_factor;
int current_delay = nSampSndCard - WebRtcAec_system_delay(aecpc->aec);
int delay_difference = 0;
@@ -848,11 +732,8 @@
current_delay += WebRtcAec_MoveFarReadPtr(aecpc->aec, 1) * PART_LEN;
}
- // We use -1 to signal an initialized state in the "extended" implementation;
- // compensate for that.
- aecpc->filtDelay = aecpc->filtDelay < 0 ? 0 : aecpc->filtDelay;
aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
- 0.2 * current_delay));
+ 0.2 * current_delay));
delay_difference = aecpc->filtDelay - aecpc->knownDelay;
if (delay_difference > 224) {
@@ -875,58 +756,6 @@
if (aecpc->timeForDelayChange > 25) {
aecpc->knownDelay = WEBRTC_SPL_MAX((int) aecpc->filtDelay - 160, 0);
}
-}
-static void EstBufDelayExtended(aecpc_t* self) {
- int reported_delay = self->msInSndCardBuf * sampMsNb * self->rate_factor;
- int current_delay = reported_delay - WebRtcAec_system_delay(self->aec);
- int delay_difference = 0;
-
- // Before we proceed with the delay estimate filtering we:
- // 1) Compensate for the frame that will be read.
- // 2) Compensate for drift resampling.
- // 3) Compensate for non-causality if needed, since the estimated delay can't
- // be negative.
-
- // 1) Compensating for the frame(s) that will be read/processed.
- current_delay += FRAME_LEN * self->rate_factor;
-
- // 2) Account for resampling frame delay.
- if (self->skewMode == kAecTrue && self->resample == kAecTrue) {
- current_delay -= kResamplingDelay;
- }
-
- // 3) Compensate for non-causality, if needed, by flushing two blocks.
- if (current_delay < PART_LEN) {
- current_delay += WebRtcAec_MoveFarReadPtr(self->aec, 2) * PART_LEN;
- }
-
- if (self->filtDelay == -1) {
- self->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay);
- } else {
- self->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.95 * self->filtDelay +
- 0.05 * current_delay));
- }
-
- delay_difference = self->filtDelay - self->knownDelay;
- if (delay_difference > 384) {
- if (self->lastDelayDiff < 128) {
- self->timeForDelayChange = 0;
- } else {
- self->timeForDelayChange++;
- }
- } else if (delay_difference < 128 && self->knownDelay > 0) {
- if (self->lastDelayDiff > 384) {
- self->timeForDelayChange = 0;
- } else {
- self->timeForDelayChange++;
- }
- } else {
- self->timeForDelayChange = 0;
- }
- self->lastDelayDiff = delay_difference;
-
- if (self->timeForDelayChange > 25) {
- self->knownDelay = WEBRTC_SPL_MAX((int) self->filtDelay - 256, 0);
- }
+ return 0;
}
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
index e939c42..1298901 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
@@ -20,6 +20,8 @@
int splitSampFreq;
int scSampFreq;
float sampFactor; // scSampRate / sampFreq
+ short autoOnOff;
+ short activity;
short skewMode;
int bufSizeStart;
int knownDelay;
@@ -37,7 +39,7 @@
short msInSndCardBuf;
short filtDelay; // Filtered delay estimate.
int timeForDelayChange;
- int startup_phase;
+ int ECstartup;
int checkBuffSize;
short lastDelayDiff;
@@ -60,8 +62,6 @@
int lastError;
- int farend_started;
-
AecCore* aec;
} aecpc_t;
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index db37f0e..97ebea3 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -128,7 +128,7 @@
for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
RenderAndCapture(kDeviceBufMs);
buffer_size += samples_per_frame_;
- if (self_->startup_phase == 0) {
+ if (self_->ECstartup == 0) {
// We have left the startup phase.
break;
}
@@ -222,7 +222,7 @@
RenderAndCapture(reported_delay_ms);
buffer_size += samples_per_frame_;
buffer_offset_ms = -buffer_offset_ms;
- if (self_->startup_phase == 0) {
+ if (self_->ECstartup == 0) {
// We have left the startup phase.
break;
}
@@ -268,7 +268,7 @@
for (; process_time_ms <= kMaxConvergenceMs; process_time_ms += 10) {
RenderAndCapture(kDeviceBufMs);
buffer_size += samples_per_frame_;
- if (self_->startup_phase == 0) {
+ if (self_->ECstartup == 0) {
// We have left the startup phase.
break;
}
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index cd12363..47ee802 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -13,14 +13,12 @@
#include <assert.h>
#include <string.h>
-extern "C" {
-#include "webrtc/modules/audio_processing/aec/aec_core.h"
-}
-#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
+
namespace webrtc {
typedef void Handle;
@@ -71,8 +69,7 @@
stream_drift_samples_(0),
was_stream_drift_set_(false),
stream_has_echo_(false),
- delay_logging_enabled_(false),
- delay_correction_enabled_(false) {}
+ delay_logging_enabled_(false) {}
EchoCancellationImpl::~EchoCancellationImpl() {}
@@ -341,11 +338,6 @@
return apm_->kNoError;
}
-void EchoCancellationImpl::SetExtraOptions(const Config& config) {
- delay_correction_enabled_ = config.Get<DelayCorrection>().enabled;
- Configure();
-}
-
void* EchoCancellationImpl::CreateHandle() const {
Handle* handle = NULL;
if (WebRtcAec_Create(&handle) != apm_->kNoError) {
@@ -377,8 +369,6 @@
config.skewMode = drift_compensation_enabled_;
config.delay_logging = delay_logging_enabled_;
- WebRtcAec_enable_delay_correction(WebRtcAec_aec_core(
- static_cast<Handle*>(handle)), delay_correction_enabled_ ? 1 : 0);
return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
}
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.h b/webrtc/modules/audio_processing/echo_cancellation_impl.h
index 2b57ebb..07506d4 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.h
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.h
@@ -15,30 +15,6 @@
namespace webrtc {
-// Use to enable the delay correction feature. This now engages an extended
-// filter mode in the AEC, along with robustness measures around the reported
-// system delays. It comes with a significant increase in AEC complexity, but is
-// much more robust to unreliable reported delays.
-//
-// Detailed changes to the algorithm:
-// - The filter length is changed from 48 to 128 ms. This comes with tuning of
-// several parameters: i) filter adaptation stepsize and error threshold;
-// ii) non-linear processing smoothing and overdrive.
-// - Option to ignore the reported delays on platforms which we deem
-// sufficiently unreliable. See WEBRTC_UNTRUSTED_DELAY in echo_cancellation.c.
-// - Faster startup times by removing the excessive "startup phase" processing
-// of reported delays.
-// - Much more conservative adjustments to the far-end read pointer. We smooth
-// the delay difference more heavily, and back off from the difference more.
-// Adjustments force a readaptation of the filter, so they should be avoided
-// except when really necessary.
-struct DelayCorrection {
- DelayCorrection() : enabled(false) {}
- DelayCorrection(bool enabled) : enabled(enabled) {}
-
- bool enabled;
-};
-
class AudioProcessingImpl;
class AudioBuffer;
@@ -58,7 +34,6 @@
// ProcessingComponent implementation.
virtual int Initialize() OVERRIDE;
- virtual void SetExtraOptions(const Config& config) OVERRIDE;
private:
// EchoCancellation implementation.
@@ -95,7 +70,6 @@
bool was_stream_drift_set_;
bool stream_has_echo_;
bool delay_logging_enabled_;
- bool delay_correction_enabled_;
};
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
deleted file mode 100644
index 16ecf02..0000000
--- a/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testing/gtest/include/gtest/gtest.h"
-extern "C" {
-#include "webrtc/modules/audio_processing/aec/aec_core.h"
-}
-#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-TEST(EchoCancellationInternalTest, DelayCorrection) {
- scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(0));
- EXPECT_TRUE(ap->echo_cancellation()->aec_core() == NULL);
-
- EXPECT_EQ(ap->kNoError, ap->echo_cancellation()->Enable(true));
- EXPECT_TRUE(ap->echo_cancellation()->is_enabled());
-
- AecCore* aec_core = ap->echo_cancellation()->aec_core();
- ASSERT_TRUE(aec_core != NULL);
- // Disabled by default.
- EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
-
- Config config;
- config.Set<DelayCorrection>(new DelayCorrection(true));
- ap->SetExtraOptions(config);
- EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
-
- // Retains setting after initialization.
- EXPECT_EQ(ap->kNoError, ap->Initialize());
- EXPECT_EQ(1, WebRtcAec_delay_correction_enabled(aec_core));
-
- config.Set<DelayCorrection>(new DelayCorrection(false));
- ap->SetExtraOptions(config);
- EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
-
- // Retains setting after initialization.
- EXPECT_EQ(ap->kNoError, ap->Initialize());
- EXPECT_EQ(0, WebRtcAec_delay_correction_enabled(aec_core));
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 5dab6f7..d5d3cbe 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -145,7 +145,6 @@
'audio_coding/neteq4/mock/mock_payload_splitter.h',
'audio_processing/aec/system_delay_unittest.cc',
'audio_processing/aec/echo_cancellation_unittest.cc',
- 'audio_processing/echo_cancellation_impl_unittest.cc',
'audio_processing/test/audio_processing_unittest.cc',
'audio_processing/utility/delay_estimator_unittest.cc',
'audio_processing/utility/ring_buffer_unittest.cc',