| /* |
| * 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 <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 "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(rtc::CriticalSection* crit_render, |
| rtc::CriticalSection* crit_capture) |
| : crit_render_(crit_render), |
| crit_capture_(crit_capture), |
| 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()) { |
| RTC_DCHECK(crit_render); |
| RTC_DCHECK(crit_capture); |
| } |
| |
| EchoCancellationImpl::~EchoCancellationImpl() = default; |
| |
| void EchoCancellationImpl::ProcessRenderAudio( |
| rtc::ArrayView<const float> packed_render_audio) { |
| rtc::CritScope cs_capture(crit_capture_); |
| 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) { |
| rtc::CritScope cs_capture(crit_capture_); |
| 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) { |
| // Run in a single-threaded manner. |
| rtc::CritScope cs_render(crit_render_); |
| rtc::CritScope cs_capture(crit_capture_); |
| |
| 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 { |
| rtc::CritScope cs(crit_capture_); |
| return enabled_; |
| } |
| |
| int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) { |
| { |
| if (MapSetting(level) == -1) { |
| return AudioProcessing::kBadParameterError; |
| } |
| rtc::CritScope cs(crit_capture_); |
| suppression_level_ = level; |
| } |
| return Configure(); |
| } |
| |
| EchoCancellationImpl::SuppressionLevel EchoCancellationImpl::suppression_level() |
| const { |
| rtc::CritScope cs(crit_capture_); |
| return suppression_level_; |
| } |
| |
| int EchoCancellationImpl::enable_drift_compensation(bool enable) { |
| { |
| rtc::CritScope cs(crit_capture_); |
| drift_compensation_enabled_ = enable; |
| } |
| return Configure(); |
| } |
| |
| bool EchoCancellationImpl::is_drift_compensation_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| return drift_compensation_enabled_; |
| } |
| |
| void EchoCancellationImpl::set_stream_drift_samples(int drift) { |
| rtc::CritScope cs(crit_capture_); |
| was_stream_drift_set_ = true; |
| stream_drift_samples_ = drift; |
| } |
| |
| int EchoCancellationImpl::stream_drift_samples() const { |
| rtc::CritScope cs(crit_capture_); |
| return stream_drift_samples_; |
| } |
| |
| int EchoCancellationImpl::enable_metrics(bool enable) { |
| { |
| rtc::CritScope cs(crit_capture_); |
| metrics_enabled_ = enable; |
| } |
| return Configure(); |
| } |
| |
| bool EchoCancellationImpl::are_metrics_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| 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) { |
| rtc::CritScope cs(crit_capture_); |
| 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 { |
| rtc::CritScope cs(crit_capture_); |
| return stream_has_echo_; |
| } |
| |
| int EchoCancellationImpl::enable_delay_logging(bool enable) { |
| { |
| rtc::CritScope cs(crit_capture_); |
| delay_logging_enabled_ = enable; |
| } |
| return Configure(); |
| } |
| |
| bool EchoCancellationImpl::is_delay_logging_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| return enabled_ && delay_logging_enabled_; |
| } |
| |
| bool EchoCancellationImpl::is_delay_agnostic_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| return delay_agnostic_enabled_; |
| } |
| |
| std::string EchoCancellationImpl::GetExperimentsDescription() { |
| rtc::CritScope cs(crit_capture_); |
| return refined_adaptive_filter_enabled_ ? "RefinedAdaptiveFilter;" : ""; |
| } |
| |
| bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| return refined_adaptive_filter_enabled_; |
| } |
| |
| bool EchoCancellationImpl::is_extended_filter_enabled() const { |
| rtc::CritScope cs(crit_capture_); |
| return extended_filter_enabled_; |
| } |
| |
| // TODO(bjornv): How should we handle the multi-channel case? |
| int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) { |
| rtc::CritScope cs(crit_capture_); |
| float fraction_poor_delays = 0; |
| return GetDelayMetrics(median, std, &fraction_poor_delays); |
| } |
| |
| int EchoCancellationImpl::GetDelayMetrics(int* median, |
| int* std, |
| float* fraction_poor_delays) { |
| rtc::CritScope cs(crit_capture_); |
| 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 { |
| rtc::CritScope cs(crit_capture_); |
| 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) { |
| rtc::CritScope cs_render(crit_render_); |
| rtc::CritScope cs_capture(crit_capture_); |
| |
| 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::CritScope cs(crit_capture_); |
| 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) { |
| { |
| rtc::CritScope cs(crit_capture_); |
| 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() { |
| rtc::CritScope cs_render(crit_render_); |
| rtc::CritScope cs_capture(crit_capture_); |
| 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 |