webrtc_m130/webrtc/modules/audio_processing/echo_cancellation_impl.cc

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/*
* 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.
*/
#include "webrtc/modules/audio_processing/echo_cancellation_impl.h"
#include <assert.h>
#include <string.h>
Add an extended filter mode to AEC. Re-land: http://review.webrtc.org/2151007/ TBR=bjornv@webrtc.org Original change description: 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 Review URL: https://webrtc-codereview.appspot.com/2295006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:17:38 +00:00
extern "C" {
#include "webrtc/modules/audio_processing/aec/aec_core.h"
}
#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
typedef void Handle;
namespace {
int16_t MapSetting(EchoCancellation::SuppressionLevel level) {
switch (level) {
case EchoCancellation::kLowSuppression:
return kAecNlpConservative;
case EchoCancellation::kModerateSuppression:
return kAecNlpModerate;
case EchoCancellation::kHighSuppression:
return kAecNlpAggressive;
}
assert(false);
return -1;
}
AudioProcessing::Error MapError(int err) {
switch (err) {
case AEC_UNSUPPORTED_FUNCTION_ERROR:
return AudioProcessing::kUnsupportedFunctionError;
case AEC_BAD_PARAMETER_ERROR:
return AudioProcessing::kBadParameterError;
case AEC_BAD_PARAMETER_WARNING:
return AudioProcessing::kBadStreamParameterWarning;
default:
// AEC_UNSPECIFIED_ERROR
// AEC_UNINITIALIZED_ERROR
// AEC_NULL_POINTER_ERROR
return AudioProcessing::kUnspecifiedError;
}
}
} // namespace
EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
CriticalSectionWrapper* crit)
: ProcessingComponent(),
apm_(apm),
crit_(crit),
drift_compensation_enabled_(false),
metrics_enabled_(false),
suppression_level_(kModerateSuppression),
stream_drift_samples_(0),
was_stream_drift_set_(false),
stream_has_echo_(false),
delay_logging_enabled_(false),
extended_filter_enabled_(false),
delay_agnostic_enabled_(false) {
}
EchoCancellationImpl::~EchoCancellationImpl() {}
int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
if (!is_component_enabled()) {
return apm_->kNoError;
}
assert(audio->num_frames_per_band() <= 160);
assert(audio->num_channels() == apm_->num_reverse_channels());
int err = apm_->kNoError;
// The ordering convention must be followed to pass to the correct AEC.
size_t handle_index = 0;
for (int i = 0; i < apm_->num_output_channels(); i++) {
for (int j = 0; j < audio->num_channels(); j++) {
Handle* my_handle = static_cast<Handle*>(handle(handle_index));
err = WebRtcAec_BufferFarend(
my_handle,
audio->split_bands_const_f(j)[kBand0To8kHz],
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
audio->num_frames_per_band());
if (err != apm_->kNoError) {
return MapError(err); // TODO(ajm): warning possible?
}
handle_index++;
}
}
return apm_->kNoError;
}
int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
if (!is_component_enabled()) {
return apm_->kNoError;
}
if (!apm_->was_stream_delay_set()) {
return apm_->kStreamParameterNotSetError;
}
if (drift_compensation_enabled_ && !was_stream_drift_set_) {
return apm_->kStreamParameterNotSetError;
}
assert(audio->num_frames_per_band() <= 160);
assert(audio->num_channels() == apm_->num_output_channels());
int err = apm_->kNoError;
// The ordering convention must be followed to pass to the correct AEC.
size_t handle_index = 0;
stream_has_echo_ = false;
for (int i = 0; i < audio->num_channels(); i++) {
for (int j = 0; j < apm_->num_reverse_channels(); j++) {
Handle* my_handle = handle(handle_index);
err = WebRtcAec_Process(
my_handle,
audio->split_bands_const_f(i),
audio->num_bands(),
audio->split_bands_f(i),
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
audio->num_frames_per_band(),
apm_->stream_delay_ms(),
stream_drift_samples_);
if (err != apm_->kNoError) {
err = MapError(err);
// TODO(ajm): Figure out how to return warnings properly.
if (err != apm_->kBadStreamParameterWarning) {
return err;
}
}
int status = 0;
err = WebRtcAec_get_echo_status(my_handle, &status);
if (err != apm_->kNoError) {
return MapError(err);
}
if (status == 1) {
stream_has_echo_ = true;
}
handle_index++;
}
}
was_stream_drift_set_ = false;
return apm_->kNoError;
}
int EchoCancellationImpl::Enable(bool enable) {
CriticalSectionScoped crit_scoped(crit_);
// Ensure AEC and AECM are not both enabled.
if (enable && apm_->echo_control_mobile()->is_enabled()) {
return apm_->kBadParameterError;
}
return EnableComponent(enable);
}
bool EchoCancellationImpl::is_enabled() const {
return is_component_enabled();
}
int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
CriticalSectionScoped crit_scoped(crit_);
if (MapSetting(level) == -1) {
return apm_->kBadParameterError;
}
suppression_level_ = level;
return Configure();
}
EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
const {
return suppression_level_;
}
int EchoCancellationImpl::enable_drift_compensation(bool enable) {
CriticalSectionScoped crit_scoped(crit_);
drift_compensation_enabled_ = enable;
return Configure();
}
bool EchoCancellationImpl::is_drift_compensation_enabled() const {
return drift_compensation_enabled_;
}
void EchoCancellationImpl::set_stream_drift_samples(int drift) {
was_stream_drift_set_ = true;
stream_drift_samples_ = drift;
}
int EchoCancellationImpl::stream_drift_samples() const {
return stream_drift_samples_;
}
int EchoCancellationImpl::enable_metrics(bool enable) {
CriticalSectionScoped crit_scoped(crit_);
metrics_enabled_ = enable;
return Configure();
}
bool EchoCancellationImpl::are_metrics_enabled() const {
return metrics_enabled_;
}
// TODO(ajm): we currently just use the metrics from the first AEC. Think more
// aboue the best way to extend this to multi-channel.
int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
CriticalSectionScoped crit_scoped(crit_);
if (metrics == NULL) {
return apm_->kNullPointerError;
}
if (!is_component_enabled() || !metrics_enabled_) {
return apm_->kNotEnabledError;
}
AecMetrics my_metrics;
memset(&my_metrics, 0, sizeof(my_metrics));
memset(metrics, 0, sizeof(Metrics));
Handle* my_handle = static_cast<Handle*>(handle(0));
int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
if (err != apm_->kNoError) {
return MapError(err);
}
metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
metrics->echo_return_loss.instant = my_metrics.erl.instant;
metrics->echo_return_loss.average = my_metrics.erl.average;
metrics->echo_return_loss.maximum = my_metrics.erl.max;
metrics->echo_return_loss.minimum = my_metrics.erl.min;
metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
metrics->a_nlp.instant = my_metrics.aNlp.instant;
metrics->a_nlp.average = my_metrics.aNlp.average;
metrics->a_nlp.maximum = my_metrics.aNlp.max;
metrics->a_nlp.minimum = my_metrics.aNlp.min;
return apm_->kNoError;
}
bool EchoCancellationImpl::stream_has_echo() const {
return stream_has_echo_;
}
int EchoCancellationImpl::enable_delay_logging(bool enable) {
CriticalSectionScoped crit_scoped(crit_);
delay_logging_enabled_ = enable;
return Configure();
}
bool EchoCancellationImpl::is_delay_logging_enabled() const {
return delay_logging_enabled_;
}
bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
return delay_agnostic_enabled_;
}
bool EchoCancellationImpl::is_extended_filter_enabled() const {
return extended_filter_enabled_;
}
// TODO(bjornv): How should we handle the multi-channel case?
int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
float fraction_poor_delays = 0;
return GetDelayMetrics(median, std, &fraction_poor_delays);
}
int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
float* fraction_poor_delays) {
CriticalSectionScoped crit_scoped(crit_);
if (median == NULL) {
return apm_->kNullPointerError;
}
if (std == NULL) {
return apm_->kNullPointerError;
}
if (!is_component_enabled() || !delay_logging_enabled_) {
return apm_->kNotEnabledError;
}
Handle* my_handle = static_cast<Handle*>(handle(0));
const int err =
WebRtcAec_GetDelayMetrics(my_handle, median, std, fraction_poor_delays);
if (err != apm_->kNoError) {
return MapError(err);
}
return apm_->kNoError;
}
struct AecCore* EchoCancellationImpl::aec_core() const {
CriticalSectionScoped crit_scoped(crit_);
if (!is_component_enabled()) {
return NULL;
}
Handle* my_handle = static_cast<Handle*>(handle(0));
return WebRtcAec_aec_core(my_handle);
}
int EchoCancellationImpl::Initialize() {
int err = ProcessingComponent::Initialize();
if (err != apm_->kNoError || !is_component_enabled()) {
return err;
}
return apm_->kNoError;
}
Add an extended filter mode to AEC. Re-land: http://review.webrtc.org/2151007/ TBR=bjornv@webrtc.org Original change description: 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 Review URL: https://webrtc-codereview.appspot.com/2295006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:17:38 +00:00
void EchoCancellationImpl::SetExtraOptions(const Config& config) {
extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
Add an extended filter mode to AEC. Re-land: http://review.webrtc.org/2151007/ TBR=bjornv@webrtc.org Original change description: 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 Review URL: https://webrtc-codereview.appspot.com/2295006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:17:38 +00:00
Configure();
}
void* EchoCancellationImpl::CreateHandle() const {
return WebRtcAec_Create();
}
void EchoCancellationImpl::DestroyHandle(void* handle) const {
assert(handle != NULL);
WebRtcAec_Free(static_cast<Handle*>(handle));
}
int EchoCancellationImpl::InitializeHandle(void* handle) const {
assert(handle != NULL);
// TODO(ajm): Drift compensation is disabled in practice. If restored, it
// should be managed internally and not depend on the hardware sample rate.
// For now, just hardcode a 48 kHz value.
return WebRtcAec_Init(static_cast<Handle*>(handle),
apm_->proc_sample_rate_hz(),
48000);
}
int EchoCancellationImpl::ConfigureHandle(void* handle) const {
assert(handle != NULL);
AecConfig config;
config.metricsMode = metrics_enabled_;
config.nlpMode = MapSetting(suppression_level_);
config.skewMode = drift_compensation_enabled_;
config.delay_logging = delay_logging_enabled_;
WebRtcAec_enable_extended_filter(
WebRtcAec_aec_core(static_cast<Handle*>(handle)),
extended_filter_enabled_ ? 1 : 0);
WebRtcAec_enable_delay_agnostic(
WebRtcAec_aec_core(static_cast<Handle*>(handle)),
delay_agnostic_enabled_ ? 1 : 0);
return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
}
int EchoCancellationImpl::num_handles_required() const {
return apm_->num_output_channels() *
apm_->num_reverse_channels();
}
int EchoCancellationImpl::GetHandleError(void* handle) const {
assert(handle != NULL);
return AudioProcessing::kUnspecifiedError;
}
} // namespace webrtc