tg2sip/webrtc_dsp/modules/audio_processing/echo_cancellation_impl.cc

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* 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 "modules/audio_processing/echo_cancellation_impl.h"
#include <stdint.h>
#include <string.h>
#include "modules/audio_processing/aec/aec_core.h"
#include "modules/audio_processing/aec/echo_cancellation.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/config.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
int16_t MapSetting(EchoCancellationImpl::SuppressionLevel level) {
switch (level) {
case EchoCancellationImpl::kLowSuppression:
return kAecNlpConservative;
case EchoCancellationImpl::kModerateSuppression:
return kAecNlpModerate;
case EchoCancellationImpl::kHighSuppression:
return kAecNlpAggressive;
}
RTC_NOTREACHED();
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;
}
}
bool EnforceZeroStreamDelay() {
#if defined(CHROMEOS)
return !field_trial::IsEnabled("WebRTC-Aec2ZeroStreamDelayKillSwitch");
#else
return false;
#endif
}
} // namespace
struct EchoCancellationImpl::StreamProperties {
StreamProperties() = delete;
StreamProperties(int sample_rate_hz,
size_t num_reverse_channels,
size_t num_output_channels,
size_t num_proc_channels)
: sample_rate_hz(sample_rate_hz),
num_reverse_channels(num_reverse_channels),
num_output_channels(num_output_channels),
num_proc_channels(num_proc_channels) {}
const int sample_rate_hz;
const size_t num_reverse_channels;
const size_t num_output_channels;
const size_t num_proc_channels;
};
class EchoCancellationImpl::Canceller {
public:
Canceller() {
state_ = WebRtcAec_Create();
RTC_DCHECK(state_);
}
~Canceller() {
RTC_CHECK(state_);
WebRtcAec_Free(state_);
}
void* state() { return state_; }
void Initialize(int sample_rate_hz) {
// 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.
const int error = WebRtcAec_Init(state_, sample_rate_hz, 48000);
RTC_DCHECK_EQ(0, error);
}
private:
void* state_;
};
EchoCancellationImpl::EchoCancellationImpl()
: drift_compensation_enabled_(false),
metrics_enabled_(true),
suppression_level_(kHighSuppression),
stream_drift_samples_(0),
was_stream_drift_set_(false),
stream_has_echo_(false),
delay_logging_enabled_(true),
extended_filter_enabled_(false),
delay_agnostic_enabled_(false),
enforce_zero_stream_delay_(EnforceZeroStreamDelay()) {}
EchoCancellationImpl::~EchoCancellationImpl() = default;
void EchoCancellationImpl::ProcessRenderAudio(
rtc::ArrayView<const float> packed_render_audio) {
if (!enabled_) {
return;
}
RTC_DCHECK(stream_properties_);
size_t handle_index = 0;
size_t buffer_index = 0;
const size_t num_frames_per_band =
packed_render_audio.size() / (stream_properties_->num_output_channels *
stream_properties_->num_reverse_channels);
for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
WebRtcAec_BufferFarend(cancellers_[handle_index++]->state(),
&packed_render_audio[buffer_index],
num_frames_per_band);
buffer_index += num_frames_per_band;
}
}
}
int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
int stream_delay_ms) {
if (!enabled_) {
return AudioProcessing::kNoError;
}
const int stream_delay_ms_use =
enforce_zero_stream_delay_ ? 0 : stream_delay_ms;
if (drift_compensation_enabled_ && !was_stream_drift_set_) {
return AudioProcessing::kStreamParameterNotSetError;
}
RTC_DCHECK(stream_properties_);
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);
int err = AudioProcessing::kNoError;
// The ordering convention must be followed to pass to the correct AEC.
size_t handle_index = 0;
stream_has_echo_ = false;
for (size_t i = 0; i < audio->num_channels(); i++) {
for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
err = WebRtcAec_Process(cancellers_[handle_index]->state(),
audio->split_bands_const_f(i), audio->num_bands(),
audio->split_bands_f(i),
audio->num_frames_per_band(), stream_delay_ms_use,
stream_drift_samples_);
if (err != AudioProcessing::kNoError) {
err = MapError(err);
// TODO(ajm): Figure out how to return warnings properly.
if (err != AudioProcessing::kBadStreamParameterWarning) {
return err;
}
}
int status = 0;
err = WebRtcAec_get_echo_status(cancellers_[handle_index]->state(),
&status);
if (err != AudioProcessing::kNoError) {
return MapError(err);
}
if (status == 1) {
stream_has_echo_ = true;
}
handle_index++;
}
}
was_stream_drift_set_ = false;
return AudioProcessing::kNoError;
}
int EchoCancellationImpl::Enable(bool enable) {
if (enable && !enabled_) {
enabled_ = enable; // Must be set before Initialize() is called.
// TODO(peah): Simplify once the Enable function has been removed from
// the public APM API.
RTC_DCHECK(stream_properties_);
Initialize(stream_properties_->sample_rate_hz,
stream_properties_->num_reverse_channels,
stream_properties_->num_output_channels,
stream_properties_->num_proc_channels);
} else {
enabled_ = enable;
}
return AudioProcessing::kNoError;
}
bool EchoCancellationImpl::is_enabled() const {
return enabled_;
}
int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
if (MapSetting(level) == -1) {
return AudioProcessing::kBadParameterError;
}
suppression_level_ = level;
return Configure();
}
EchoCancellationImpl::SuppressionLevel EchoCancellationImpl::suppression_level()
const {
return suppression_level_;
}
int EchoCancellationImpl::enable_drift_compensation(bool enable) {
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) {
metrics_enabled_ = enable;
return Configure();
}
bool EchoCancellationImpl::are_metrics_enabled() const {
return enabled_ && 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) {
if (metrics == NULL) {
return AudioProcessing::kNullPointerError;
}
if (!enabled_ || !metrics_enabled_) {
return AudioProcessing::kNotEnabledError;
}
AecMetrics my_metrics;
memset(&my_metrics, 0, sizeof(my_metrics));
memset(metrics, 0, sizeof(Metrics));
const int err = WebRtcAec_GetMetrics(cancellers_[0]->state(), &my_metrics);
if (err != AudioProcessing::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;
metrics->divergent_filter_fraction = my_metrics.divergent_filter_fraction;
return AudioProcessing::kNoError;
}
bool EchoCancellationImpl::stream_has_echo() const {
return stream_has_echo_;
}
int EchoCancellationImpl::enable_delay_logging(bool enable) {
delay_logging_enabled_ = enable;
return Configure();
}
bool EchoCancellationImpl::is_delay_logging_enabled() const {
return enabled_ && delay_logging_enabled_;
}
bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
return delay_agnostic_enabled_;
}
std::string EchoCancellationImpl::GetExperimentsDescription() {
return refined_adaptive_filter_enabled_ ? "RefinedAdaptiveFilter;" : "";
}
bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const {
return refined_adaptive_filter_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) {
if (median == NULL) {
return AudioProcessing::kNullPointerError;
}
if (std == NULL) {
return AudioProcessing::kNullPointerError;
}
if (!enabled_ || !delay_logging_enabled_) {
return AudioProcessing::kNotEnabledError;
}
const int err = WebRtcAec_GetDelayMetrics(cancellers_[0]->state(), median,
std, fraction_poor_delays);
if (err != AudioProcessing::kNoError) {
return MapError(err);
}
return AudioProcessing::kNoError;
}
struct AecCore* EchoCancellationImpl::aec_core() const {
if (!enabled_) {
return NULL;
}
return WebRtcAec_aec_core(cancellers_[0]->state());
}
void EchoCancellationImpl::Initialize(int sample_rate_hz,
size_t num_reverse_channels,
size_t num_output_channels,
size_t num_proc_channels) {
stream_properties_.reset(
new StreamProperties(sample_rate_hz, num_reverse_channels,
num_output_channels, num_proc_channels));
if (!enabled_) {
return;
}
const size_t num_cancellers_required =
NumCancellersRequired(stream_properties_->num_output_channels,
stream_properties_->num_reverse_channels);
if (num_cancellers_required > cancellers_.size()) {
const size_t cancellers_old_size = cancellers_.size();
cancellers_.resize(num_cancellers_required);
for (size_t i = cancellers_old_size; i < cancellers_.size(); ++i) {
cancellers_[i].reset(new Canceller());
}
}
for (auto& canceller : cancellers_) {
canceller->Initialize(sample_rate_hz);
}
Configure();
}
int EchoCancellationImpl::GetSystemDelayInSamples() const {
RTC_DCHECK(enabled_);
// Report the delay for the first AEC component.
return WebRtcAec_system_delay(WebRtcAec_aec_core(cancellers_[0]->state()));
}
void EchoCancellationImpl::PackRenderAudioBuffer(
const AudioBuffer* audio,
size_t num_output_channels,
size_t num_channels,
std::vector<float>* packed_buffer) {
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(num_channels, audio->num_channels());
packed_buffer->clear();
// The ordering convention must be followed to pass the correct data.
for (size_t i = 0; i < num_output_channels; i++) {
for (size_t j = 0; j < audio->num_channels(); j++) {
// Buffer the samples in the render queue.
packed_buffer->insert(packed_buffer->end(),
audio->split_bands_const_f(j)[kBand0To8kHz],
(audio->split_bands_const_f(j)[kBand0To8kHz] +
audio->num_frames_per_band()));
}
}
}
void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
{
extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
refined_adaptive_filter_enabled_ =
config.Get<RefinedAdaptiveFilter>().enabled;
}
Configure();
}
int EchoCancellationImpl::Configure() {
AecConfig config;
config.metricsMode = metrics_enabled_;
config.nlpMode = MapSetting(suppression_level_);
config.skewMode = drift_compensation_enabled_;
config.delay_logging = delay_logging_enabled_;
int error = AudioProcessing::kNoError;
for (auto& canceller : cancellers_) {
WebRtcAec_enable_extended_filter(WebRtcAec_aec_core(canceller->state()),
extended_filter_enabled_ ? 1 : 0);
WebRtcAec_enable_delay_agnostic(WebRtcAec_aec_core(canceller->state()),
delay_agnostic_enabled_ ? 1 : 0);
WebRtcAec_enable_refined_adaptive_filter(
WebRtcAec_aec_core(canceller->state()),
refined_adaptive_filter_enabled_);
const int handle_error = WebRtcAec_set_config(canceller->state(), config);
if (handle_error != AudioProcessing::kNoError) {
error = AudioProcessing::kNoError;
}
}
return error;
}
size_t EchoCancellationImpl::NumCancellersRequired(
size_t num_output_channels,
size_t num_reverse_channels) {
return num_output_channels * num_reverse_channels;
}
} // namespace webrtc