blob: 864db5308e62aecf7b335889dcd9283687366b57 [file] [log] [blame]
/*
* Copyright (c) 2012 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.
*/
/*
* Contains the API functions for the AEC.
*/
#include "modules/audio_processing/aec/echo_cancellation.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
extern "C" {
#include "common_audio/ring_buffer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
}
#include "modules/audio_processing/aec/aec_core.h"
#include "modules/audio_processing/aec/aec_resampler.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
Aec::Aec() = default;
Aec::~Aec() = default;
// 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_UNTRUSTED_DELAY) && defined(WEBRTC_MAC)
static const int kDelayDiffOffsetSamples = -160;
#else
// Not enabled for now.
static const int kDelayDiffOffsetSamples = 0;
#endif
#if defined(WEBRTC_MAC)
static const int kFixedDelayMs = 20;
#else
static const int kFixedDelayMs = 50;
#endif
#if !defined(WEBRTC_UNTRUSTED_DELAY)
static const int kMinTrustedDelayMs = 20;
#endif
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
// TODO(bjornv): Replace with kResamplerBufferSize
#define MAX_RESAMP_LEN (5 * FRAME_LEN)
static const int kMaxBufSizeStart = 62; // In partitions
static const int sampMsNb = 8; // samples per ms in nb
static const int initCheck = 42;
int Aec::instance_count = 0;
// Estimates delay to set the position of the far-end buffer read pointer
// (controlled by knownDelay)
static void EstBufDelayNormal(Aec* aecInst);
static void EstBufDelayExtended(Aec* aecInst);
static int ProcessNormal(Aec* aecInst,
const float* const* nearend,
size_t num_bands,
float* const* out,
size_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
static void ProcessExtended(Aec* aecInst,
const float* const* nearend,
size_t num_bands,
float* const* out,
size_t num_samples,
int16_t reported_delay_ms,
int32_t skew);
void* WebRtcAec_Create() {
Aec* aecpc = new Aec();
if (!aecpc) {
return NULL;
}
aecpc->data_dumper.reset(new ApmDataDumper(aecpc->instance_count));
aecpc->aec = WebRtcAec_CreateAec(aecpc->instance_count);
if (!aecpc->aec) {
WebRtcAec_Free(aecpc);
return NULL;
}
aecpc->resampler = WebRtcAec_CreateResampler();
if (!aecpc->resampler) {
WebRtcAec_Free(aecpc);
return NULL;
}
// Create far-end pre-buffer. The buffer size has to be large enough for
// largest possible drift compensation (kResamplerBufferSize) + "almost" an
// FFT buffer (PART_LEN2 - 1).
aecpc->far_pre_buf =
WebRtc_CreateBuffer(PART_LEN2 + kResamplerBufferSize, sizeof(float));
if (!aecpc->far_pre_buf) {
WebRtcAec_Free(aecpc);
return NULL;
}
aecpc->initFlag = 0;
aecpc->instance_count++;
return aecpc;
}
void WebRtcAec_Free(void* aecInst) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
if (aecpc == NULL) {
return;
}
WebRtc_FreeBuffer(aecpc->far_pre_buf);
WebRtcAec_FreeAec(aecpc->aec);
WebRtcAec_FreeResampler(aecpc->resampler);
delete aecpc;
}
int32_t WebRtcAec_Init(void* aecInst, int32_t sampFreq, int32_t scSampFreq) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
aecpc->data_dumper->InitiateNewSetOfRecordings();
AecConfig aecConfig;
if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000 &&
sampFreq != 48000) {
return AEC_BAD_PARAMETER_ERROR;
}
aecpc->sampFreq = sampFreq;
if (scSampFreq < 1 || scSampFreq > 96000) {
return AEC_BAD_PARAMETER_ERROR;
}
aecpc->scSampFreq = scSampFreq;
// Initialize echo canceller core
if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
return AEC_UNSPECIFIED_ERROR;
}
if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
return AEC_UNSPECIFIED_ERROR;
}
WebRtc_InitBuffer(aecpc->far_pre_buf);
WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap.
aecpc->initFlag = initCheck; // indicates that initialization has been done
if (aecpc->sampFreq == 32000 || aecpc->sampFreq == 48000) {
aecpc->splitSampFreq = 16000;
} else {
aecpc->splitSampFreq = sampFreq;
}
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;
// We skip the startup_phase completely (setting to 0) if DA-AEC is enabled,
// but not extended_filter mode.
aecpc->startup_phase = WebRtcAec_extended_filter_enabled(aecpc->aec) ||
!WebRtcAec_delay_agnostic_enabled(aecpc->aec);
aecpc->bufSizeStart = 0;
aecpc->checkBufSizeCtr = 0;
aecpc->msInSndCardBuf = 0;
aecpc->filtDelay = -1; // -1 indicates an initialized state.
aecpc->timeForDelayChange = 0;
aecpc->knownDelay = 0;
aecpc->lastDelayDiff = 0;
aecpc->skewFrCtr = 0;
aecpc->resample = kAecFalse;
aecpc->highSkewCtr = 0;
aecpc->skew = 0;
aecpc->farend_started = 0;
// Default settings.
aecConfig.nlpMode = kAecNlpModerate;
aecConfig.skewMode = kAecFalse;
aecConfig.metricsMode = kAecFalse;
aecConfig.delay_logging = kAecFalse;
if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
return AEC_UNSPECIFIED_ERROR;
}
return 0;
}
// Returns any error that is caused when buffering the
// far-end signal.
int32_t WebRtcAec_GetBufferFarendError(void* aecInst,
const float* farend,
size_t nrOfSamples) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
if (!farend)
return AEC_NULL_POINTER_ERROR;
if (aecpc->initFlag != initCheck)
return AEC_UNINITIALIZED_ERROR;
// number of samples == 160 for SWB input
if (nrOfSamples != 80 && nrOfSamples != 160)
return AEC_BAD_PARAMETER_ERROR;
return 0;
}
// only buffer L band for farend
int32_t WebRtcAec_BufferFarend(void* aecInst,
const float* farend,
size_t nrOfSamples) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
size_t newNrOfSamples = nrOfSamples;
float new_farend[MAX_RESAMP_LEN];
const float* farend_ptr = farend;
// Get any error caused by buffering the farend signal.
int32_t error_code =
WebRtcAec_GetBufferFarendError(aecInst, farend, nrOfSamples);
if (error_code != 0)
return error_code;
if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
// Resample and get a new number of samples
WebRtcAec_ResampleLinear(aecpc->resampler, farend, nrOfSamples, aecpc->skew,
new_farend, &newNrOfSamples);
farend_ptr = new_farend;
}
aecpc->farend_started = 1;
WebRtcAec_SetSystemDelay(aecpc->aec, WebRtcAec_system_delay(aecpc->aec) +
static_cast<int>(newNrOfSamples));
// Write the time-domain data to |far_pre_buf|.
WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, newNrOfSamples);
// TODO(minyue): reduce to |PART_LEN| samples for each buffering.
while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
// We have enough data to pass to the FFT, hence read PART_LEN2 samples.
{
float* ptmp = NULL;
float tmp[PART_LEN2];
WebRtc_ReadBuffer(aecpc->far_pre_buf,
reinterpret_cast<void**>(&ptmp), tmp, PART_LEN2);
WebRtcAec_BufferFarendBlock(aecpc->aec, &ptmp[PART_LEN]);
}
// Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN);
}
return 0;
}
int32_t WebRtcAec_Process(void* aecInst,
const float* const* nearend,
size_t num_bands,
float* const* out,
size_t nrOfSamples,
int16_t msInSndCardBuf,
int32_t skew) {
Aec* aecpc = reinterpret_cast<Aec*>(aecInst);
int32_t retVal = 0;
if (out == NULL) {
return AEC_NULL_POINTER_ERROR;
}
if (aecpc->initFlag != initCheck) {
return AEC_UNINITIALIZED_ERROR;
}
// number of samples == 160 for SWB input
if (nrOfSamples != 80 && nrOfSamples != 160) {
return AEC_BAD_PARAMETER_ERROR;
}
if (msInSndCardBuf < 0) {
msInSndCardBuf = 0;
retVal = AEC_BAD_PARAMETER_WARNING;
} else if (msInSndCardBuf > kMaxTrustedDelayMs) {
// The clamping is now done in ProcessExtended/Normal().
retVal = AEC_BAD_PARAMETER_WARNING;
}
// This returns the value of aec->extended_filter_enabled.
if (WebRtcAec_extended_filter_enabled(aecpc->aec)) {
ProcessExtended(aecpc, nearend, num_bands, out, nrOfSamples, msInSndCardBuf,
skew);
} else {
retVal = ProcessNormal(aecpc, nearend, num_bands, out, nrOfSamples,
msInSndCardBuf, skew);
}
int far_buf_size_samples = WebRtcAec_system_delay(aecpc->aec);
aecpc->data_dumper->DumpRaw("aec_system_delay", 1, &far_buf_size_samples);
aecpc->data_dumper->DumpRaw("aec_known_delay", 1, &aecpc->knownDelay);
return retVal;
}
int WebRtcAec_set_config(void* handle, AecConfig config) {
Aec* self = reinterpret_cast<Aec*>(handle);
if (self->initFlag != initCheck) {
return AEC_UNINITIALIZED_ERROR;
}
if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
return AEC_BAD_PARAMETER_ERROR;
}
self->skewMode = config.skewMode;
if (config.nlpMode != kAecNlpConservative &&
config.nlpMode != kAecNlpModerate &&
config.nlpMode != kAecNlpAggressive) {
return AEC_BAD_PARAMETER_ERROR;
}
if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
return AEC_BAD_PARAMETER_ERROR;
}
if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
return AEC_BAD_PARAMETER_ERROR;
}
WebRtcAec_SetConfigCore(self->aec, config.nlpMode, config.metricsMode,
config.delay_logging);
return 0;
}
int WebRtcAec_get_echo_status(void* handle, int* status) {
Aec* self = reinterpret_cast<Aec*>(handle);
if (status == NULL) {
return AEC_NULL_POINTER_ERROR;
}
if (self->initFlag != initCheck) {
return AEC_UNINITIALIZED_ERROR;
}
*status = WebRtcAec_echo_state(self->aec);
return 0;
}
int WebRtcAec_GetMetrics(void* handle, AecMetrics* metrics) {
const float kUpWeight = 0.7f;
float dtmp;
int stmp;
Aec* self = reinterpret_cast<Aec*>(handle);
Stats erl;
Stats erle;
Stats a_nlp;
if (handle == NULL) {
return -1;
}
if (metrics == NULL) {
return AEC_NULL_POINTER_ERROR;
}
if (self->initFlag != initCheck) {
return AEC_UNINITIALIZED_ERROR;
}
WebRtcAec_GetEchoStats(self->aec, &erl, &erle, &a_nlp,
&metrics->divergent_filter_fraction);
// ERL
metrics->erl.instant = static_cast<int>(erl.instant);
if ((erl.himean > kOffsetLevel) && (erl.average > kOffsetLevel)) {
// Use a mix between regular average and upper part average.
dtmp = kUpWeight * erl.himean + (1 - kUpWeight) * erl.average;
metrics->erl.average = static_cast<int>(dtmp);
} else {
metrics->erl.average = kOffsetLevel;
}
metrics->erl.max = static_cast<int>(erl.max);
if (erl.min < (kOffsetLevel * (-1))) {
metrics->erl.min = static_cast<int>(erl.min);
} else {
metrics->erl.min = kOffsetLevel;
}
// ERLE
metrics->erle.instant = static_cast<int>(erle.instant);
if ((erle.himean > kOffsetLevel) && (erle.average > kOffsetLevel)) {
// Use a mix between regular average and upper part average.
dtmp = kUpWeight * erle.himean + (1 - kUpWeight) * erle.average;
metrics->erle.average = static_cast<int>(dtmp);
} else {
metrics->erle.average = kOffsetLevel;
}
metrics->erle.max = static_cast<int>(erle.max);
if (erle.min < (kOffsetLevel * (-1))) {
metrics->erle.min = static_cast<int>(erle.min);
} else {
metrics->erle.min = kOffsetLevel;
}
// RERL
if ((metrics->erl.average > kOffsetLevel) &&
(metrics->erle.average > kOffsetLevel)) {
stmp = metrics->erl.average + metrics->erle.average;
} else {
stmp = kOffsetLevel;
}
metrics->rerl.average = stmp;
// No other statistics needed, but returned for completeness.
metrics->rerl.instant = stmp;
metrics->rerl.max = stmp;
metrics->rerl.min = stmp;
// A_NLP
metrics->aNlp.instant = static_cast<int>(a_nlp.instant);
if ((a_nlp.himean > kOffsetLevel) && (a_nlp.average > kOffsetLevel)) {
// Use a mix between regular average and upper part average.
dtmp = kUpWeight * a_nlp.himean + (1 - kUpWeight) * a_nlp.average;
metrics->aNlp.average = static_cast<int>(dtmp);
} else {
metrics->aNlp.average = kOffsetLevel;
}
metrics->aNlp.max = static_cast<int>(a_nlp.max);
if (a_nlp.min < (kOffsetLevel * (-1))) {
metrics->aNlp.min = static_cast<int>(a_nlp.min);
} else {
metrics->aNlp.min = kOffsetLevel;
}
return 0;
}
int WebRtcAec_GetDelayMetrics(void* handle,
int* median,
int* std,
float* fraction_poor_delays) {
Aec* self = reinterpret_cast<Aec*>(handle);
if (median == NULL) {
return AEC_NULL_POINTER_ERROR;
}
if (std == NULL) {
return AEC_NULL_POINTER_ERROR;
}
if (self->initFlag != initCheck) {
return AEC_UNINITIALIZED_ERROR;
}
if (WebRtcAec_GetDelayMetricsCore(self->aec, median, std,
fraction_poor_delays) == -1) {
// Logging disabled.
return AEC_UNSUPPORTED_FUNCTION_ERROR;
}
return 0;
}
AecCore* WebRtcAec_aec_core(void* handle) {
if (!handle) {
return NULL;
}
return reinterpret_cast<Aec*>(handle)->aec;
}
static int ProcessNormal(Aec* aecInst,
const float* const* nearend,
size_t num_bands,
float* const* out,
size_t num_samples,
int16_t reported_delay_ms,
int32_t skew) {
int retVal = 0;
size_t i;
size_t nBlocks10ms;
// Limit resampling to doubling/halving of signal
const float minSkewEst = -0.5f;
const float maxSkewEst = 1.0f;
reported_delay_ms =
reported_delay_ms > kMaxTrustedDelayMs ? kMaxTrustedDelayMs :
reported_delay_ms;
// TODO(andrew): we need to investigate if this +10 is really wanted.
reported_delay_ms += 10;
aecInst->msInSndCardBuf = reported_delay_ms;
if (aecInst->skewMode == kAecTrue) {
if (aecInst->skewFrCtr < 25) {
aecInst->skewFrCtr++;
} else {
retVal = WebRtcAec_GetSkew(aecInst->resampler, skew, &aecInst->skew);
if (retVal == -1) {
aecInst->skew = 0;
retVal = AEC_BAD_PARAMETER_WARNING;
}
aecInst->skew /= aecInst->sampFactor * num_samples;
if (aecInst->skew < 1.0e-3 && aecInst->skew > -1.0e-3) {
aecInst->resample = kAecFalse;
} else {
aecInst->resample = kAecTrue;
}
if (aecInst->skew < minSkewEst) {
aecInst->skew = minSkewEst;
} else if (aecInst->skew > maxSkewEst) {
aecInst->skew = maxSkewEst;
}
aecInst->data_dumper->DumpRaw("aec_skew", 1, &aecInst->skew);
}
}
nBlocks10ms = num_samples / (FRAME_LEN * aecInst->rate_factor);
if (aecInst->startup_phase) {
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]) * num_samples);
}
}
// 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 (aecInst->checkBuffSize) {
aecInst->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 (aecInst->counter == 0) {
aecInst->firstVal = aecInst->msInSndCardBuf;
aecInst->sum = 0;
}
if (abs(aecInst->firstVal - aecInst->msInSndCardBuf) <
WEBRTC_SPL_MAX(0.2 * aecInst->msInSndCardBuf, sampMsNb)) {
aecInst->sum += aecInst->msInSndCardBuf;
aecInst->counter++;
} else {
aecInst->counter = 0;
}
if (aecInst->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.
aecInst->bufSizeStart =
WEBRTC_SPL_MIN((3 * aecInst->sum * aecInst->rate_factor * 8) /
(4 * aecInst->counter * PART_LEN),
kMaxBufSizeStart);
// Buffer size has now been determined.
aecInst->checkBuffSize = 0;
}
if (aecInst->checkBufSizeCtr * nBlocks10ms > 50) {
// For really bad systems, don't disable the echo canceller for
// more than 0.5 sec.
aecInst->bufSizeStart = WEBRTC_SPL_MIN(
(aecInst->msInSndCardBuf * aecInst->rate_factor * 3) / 40,
kMaxBufSizeStart);
aecInst->checkBuffSize = 0;
}
}
// If |checkBuffSize| changed in the if-statement above.
if (!aecInst->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(aecInst->aec) / PART_LEN -
aecInst->bufSizeStart;
if (overhead_elements == 0) {
// Enable the AEC
aecInst->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_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec,
overhead_elements);
// Enable the AEC
aecInst->startup_phase = 0;
}
}
} else {
// AEC is enabled.
EstBufDelayNormal(aecInst);
// Call the AEC.
// TODO(bjornv): Re-structure such that we don't have to pass
// |aecInst->knownDelay| as input. Change name to something like
// |system_buffer_diff|.
WebRtcAec_ProcessFrames(aecInst->aec, nearend, num_bands, num_samples,
aecInst->knownDelay, out);
}
return retVal;
}
static void ProcessExtended(Aec* self,
const float* const* near,
size_t num_bands,
float* const* out,
size_t num_samples,
int16_t reported_delay_ms,
int32_t skew) {
size_t i;
const int delay_diff_offset = kDelayDiffOffsetSamples;
RTC_DCHECK(num_samples == 80 || num_samples == 160);
#if defined(WEBRTC_UNTRUSTED_DELAY)
reported_delay_ms = kFixedDelayMs;
#else
// This is the usual mode where we trust the reported system delay values.
// 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) {
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;
}
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;
#if defined(WEBRTC_ANDROID)
int target_delay = startup_size_ms * self->rate_factor * 8;
#else
// To avoid putting the AEC in a non-causal state we're being slightly
// conservative and scale by 2. On Android we use a fixed delay and
// therefore there is no need to scale the target_delay.
int target_delay = startup_size_ms * self->rate_factor * 8 / 2;
#endif
int overhead_elements =
(WebRtcAec_system_delay(self->aec) - target_delay) / PART_LEN;
WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(self->aec,
overhead_elements);
self->startup_phase = 0;
}
EstBufDelayExtended(self);
{
// |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|.
const int adjusted_known_delay =
WEBRTC_SPL_MAX(0, self->knownDelay + delay_diff_offset);
WebRtcAec_ProcessFrames(self->aec, near, num_bands, num_samples,
adjusted_known_delay, out);
}
}
static void EstBufDelayNormal(Aec* aecInst) {
int nSampSndCard = aecInst->msInSndCardBuf * sampMsNb * aecInst->rate_factor;
int current_delay = nSampSndCard - WebRtcAec_system_delay(aecInst->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 * aecInst->rate_factor;
// 2) Account for resampling frame delay.
if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) {
current_delay -= kResamplingDelay;
}
// 3) Compensate for non-causality, if needed, by flushing one block.
if (current_delay < PART_LEN) {
current_delay +=
WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 1) *
PART_LEN;
}
// We use -1 to signal an initialized state in the "extended" implementation;
// compensate for that.
aecInst->filtDelay = aecInst->filtDelay < 0 ? 0 : aecInst->filtDelay;
aecInst->filtDelay =
WEBRTC_SPL_MAX(0, static_cast<int16_t>(0.8 *
aecInst->filtDelay +
0.2 * current_delay));
delay_difference = aecInst->filtDelay - aecInst->knownDelay;
if (delay_difference > 224) {
if (aecInst->lastDelayDiff < 96) {
aecInst->timeForDelayChange = 0;
} else {
aecInst->timeForDelayChange++;
}
} else if (delay_difference < 96 && aecInst->knownDelay > 0) {
if (aecInst->lastDelayDiff > 224) {
aecInst->timeForDelayChange = 0;
} else {
aecInst->timeForDelayChange++;
}
} else {
aecInst->timeForDelayChange = 0;
}
aecInst->lastDelayDiff = delay_difference;
if (aecInst->timeForDelayChange > 25) {
aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 160, 0);
}
}
static void EstBufDelayExtended(Aec* aecInst) {
int reported_delay = aecInst->msInSndCardBuf * sampMsNb *
aecInst->rate_factor;
int current_delay = reported_delay - WebRtcAec_system_delay(aecInst->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 * aecInst->rate_factor;
// 2) Account for resampling frame delay.
if (aecInst->skewMode == kAecTrue && aecInst->resample == kAecTrue) {
current_delay -= kResamplingDelay;
}
// 3) Compensate for non-causality, if needed, by flushing two blocks.
if (current_delay < PART_LEN) {
current_delay +=
WebRtcAec_AdjustFarendBufferSizeAndSystemDelay(aecInst->aec, 2) *
PART_LEN;
}
if (aecInst->filtDelay == -1) {
aecInst->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay);
} else {
aecInst->filtDelay = WEBRTC_SPL_MAX(
0, static_cast<int16_t>(0.95 * aecInst->filtDelay + 0.05 *
current_delay));
}
delay_difference = aecInst->filtDelay - aecInst->knownDelay;
if (delay_difference > 384) {
if (aecInst->lastDelayDiff < 128) {
aecInst->timeForDelayChange = 0;
} else {
aecInst->timeForDelayChange++;
}
} else if (delay_difference < 128 && aecInst->knownDelay > 0) {
if (aecInst->lastDelayDiff > 384) {
aecInst->timeForDelayChange = 0;
} else {
aecInst->timeForDelayChange++;
}
} else {
aecInst->timeForDelayChange = 0;
}
aecInst->lastDelayDiff = delay_difference;
if (aecInst->timeForDelayChange > 25) {
aecInst->knownDelay = WEBRTC_SPL_MAX((int)aecInst->filtDelay - 256, 0);
}
}
} // namespace webrtc