mirror of
https://github.com/danog/libtgvoip.git
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471 lines
15 KiB
C++
471 lines
15 KiB
C++
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/echo_cancellation_impl.h"
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#include <stdint.h>
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#include <string.h>
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#include "modules/audio_processing/aec/aec_core.h"
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#include "modules/audio_processing/aec/echo_cancellation.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/config.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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int16_t MapSetting(EchoCancellationImpl::SuppressionLevel level) {
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switch (level) {
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case EchoCancellationImpl::kLowSuppression:
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return kAecNlpConservative;
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case EchoCancellationImpl::kModerateSuppression:
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return kAecNlpModerate;
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case EchoCancellationImpl::kHighSuppression:
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return kAecNlpAggressive;
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}
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RTC_NOTREACHED();
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return -1;
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}
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AudioProcessing::Error MapError(int err) {
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switch (err) {
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case AEC_UNSUPPORTED_FUNCTION_ERROR:
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return AudioProcessing::kUnsupportedFunctionError;
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case AEC_BAD_PARAMETER_ERROR:
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return AudioProcessing::kBadParameterError;
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case AEC_BAD_PARAMETER_WARNING:
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return AudioProcessing::kBadStreamParameterWarning;
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default:
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// AEC_UNSPECIFIED_ERROR
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// AEC_UNINITIALIZED_ERROR
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// AEC_NULL_POINTER_ERROR
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return AudioProcessing::kUnspecifiedError;
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}
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}
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bool EnforceZeroStreamDelay() {
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#if defined(CHROMEOS)
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return !field_trial::IsEnabled("WebRTC-Aec2ZeroStreamDelayKillSwitch");
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#else
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return false;
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#endif
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}
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} // namespace
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struct EchoCancellationImpl::StreamProperties {
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StreamProperties() = delete;
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StreamProperties(int sample_rate_hz,
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size_t num_reverse_channels,
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size_t num_output_channels,
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size_t num_proc_channels)
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: sample_rate_hz(sample_rate_hz),
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num_reverse_channels(num_reverse_channels),
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num_output_channels(num_output_channels),
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num_proc_channels(num_proc_channels) {}
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const int sample_rate_hz;
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const size_t num_reverse_channels;
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const size_t num_output_channels;
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const size_t num_proc_channels;
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};
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class EchoCancellationImpl::Canceller {
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public:
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Canceller() {
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state_ = WebRtcAec_Create();
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RTC_DCHECK(state_);
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}
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~Canceller() {
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RTC_CHECK(state_);
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WebRtcAec_Free(state_);
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}
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void* state() { return state_; }
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void Initialize(int sample_rate_hz) {
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// TODO(ajm): Drift compensation is disabled in practice. If restored, it
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// should be managed internally and not depend on the hardware sample rate.
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// For now, just hardcode a 48 kHz value.
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const int error = WebRtcAec_Init(state_, sample_rate_hz, 48000);
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RTC_DCHECK_EQ(0, error);
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}
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private:
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void* state_;
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};
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EchoCancellationImpl::EchoCancellationImpl()
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: drift_compensation_enabled_(false),
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metrics_enabled_(true),
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suppression_level_(kHighSuppression),
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stream_drift_samples_(0),
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was_stream_drift_set_(false),
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stream_has_echo_(false),
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delay_logging_enabled_(true),
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extended_filter_enabled_(false),
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delay_agnostic_enabled_(false),
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enforce_zero_stream_delay_(EnforceZeroStreamDelay()) {}
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EchoCancellationImpl::~EchoCancellationImpl() = default;
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void EchoCancellationImpl::ProcessRenderAudio(
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rtc::ArrayView<const float> packed_render_audio) {
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if (!enabled_) {
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return;
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}
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RTC_DCHECK(stream_properties_);
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size_t handle_index = 0;
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size_t buffer_index = 0;
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const size_t num_frames_per_band =
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packed_render_audio.size() / (stream_properties_->num_output_channels *
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stream_properties_->num_reverse_channels);
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for (size_t i = 0; i < stream_properties_->num_output_channels; i++) {
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for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
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WebRtcAec_BufferFarend(cancellers_[handle_index++]->state(),
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&packed_render_audio[buffer_index],
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num_frames_per_band);
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buffer_index += num_frames_per_band;
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}
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}
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}
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int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
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int stream_delay_ms) {
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if (!enabled_) {
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return AudioProcessing::kNoError;
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}
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const int stream_delay_ms_use =
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enforce_zero_stream_delay_ ? 0 : stream_delay_ms;
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if (drift_compensation_enabled_ && !was_stream_drift_set_) {
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return AudioProcessing::kStreamParameterNotSetError;
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}
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RTC_DCHECK(stream_properties_);
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RTC_DCHECK_GE(160, audio->num_frames_per_band());
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RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);
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int err = AudioProcessing::kNoError;
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// The ordering convention must be followed to pass to the correct AEC.
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size_t handle_index = 0;
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stream_has_echo_ = false;
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for (size_t i = 0; i < audio->num_channels(); i++) {
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for (size_t j = 0; j < stream_properties_->num_reverse_channels; j++) {
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err = WebRtcAec_Process(cancellers_[handle_index]->state(),
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audio->split_bands_const_f(i), audio->num_bands(),
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audio->split_bands_f(i),
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audio->num_frames_per_band(), stream_delay_ms_use,
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stream_drift_samples_);
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if (err != AudioProcessing::kNoError) {
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err = MapError(err);
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// TODO(ajm): Figure out how to return warnings properly.
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if (err != AudioProcessing::kBadStreamParameterWarning) {
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return err;
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}
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}
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int status = 0;
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err = WebRtcAec_get_echo_status(cancellers_[handle_index]->state(),
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&status);
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if (err != AudioProcessing::kNoError) {
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return MapError(err);
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}
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if (status == 1) {
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stream_has_echo_ = true;
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}
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handle_index++;
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}
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}
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was_stream_drift_set_ = false;
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return AudioProcessing::kNoError;
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}
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int EchoCancellationImpl::Enable(bool enable) {
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if (enable && !enabled_) {
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enabled_ = enable; // Must be set before Initialize() is called.
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// TODO(peah): Simplify once the Enable function has been removed from
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// the public APM API.
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RTC_DCHECK(stream_properties_);
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Initialize(stream_properties_->sample_rate_hz,
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stream_properties_->num_reverse_channels,
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stream_properties_->num_output_channels,
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stream_properties_->num_proc_channels);
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} else {
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enabled_ = enable;
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}
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return AudioProcessing::kNoError;
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}
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bool EchoCancellationImpl::is_enabled() const {
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return enabled_;
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}
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int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
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if (MapSetting(level) == -1) {
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return AudioProcessing::kBadParameterError;
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}
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suppression_level_ = level;
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return Configure();
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}
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EchoCancellationImpl::SuppressionLevel EchoCancellationImpl::suppression_level()
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const {
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return suppression_level_;
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}
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int EchoCancellationImpl::enable_drift_compensation(bool enable) {
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drift_compensation_enabled_ = enable;
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return Configure();
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}
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bool EchoCancellationImpl::is_drift_compensation_enabled() const {
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return drift_compensation_enabled_;
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}
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void EchoCancellationImpl::set_stream_drift_samples(int drift) {
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was_stream_drift_set_ = true;
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stream_drift_samples_ = drift;
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}
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int EchoCancellationImpl::stream_drift_samples() const {
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return stream_drift_samples_;
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}
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int EchoCancellationImpl::enable_metrics(bool enable) {
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metrics_enabled_ = enable;
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return Configure();
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}
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bool EchoCancellationImpl::are_metrics_enabled() const {
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return enabled_ && metrics_enabled_;
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}
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// TODO(ajm): we currently just use the metrics from the first AEC. Think more
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// aboue the best way to extend this to multi-channel.
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int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
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if (metrics == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (!enabled_ || !metrics_enabled_) {
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return AudioProcessing::kNotEnabledError;
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}
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AecMetrics my_metrics;
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memset(&my_metrics, 0, sizeof(my_metrics));
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memset(metrics, 0, sizeof(Metrics));
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const int err = WebRtcAec_GetMetrics(cancellers_[0]->state(), &my_metrics);
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if (err != AudioProcessing::kNoError) {
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return MapError(err);
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}
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metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
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metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
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metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
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metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
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metrics->echo_return_loss.instant = my_metrics.erl.instant;
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metrics->echo_return_loss.average = my_metrics.erl.average;
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metrics->echo_return_loss.maximum = my_metrics.erl.max;
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metrics->echo_return_loss.minimum = my_metrics.erl.min;
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metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
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metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
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metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
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metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
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metrics->a_nlp.instant = my_metrics.aNlp.instant;
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metrics->a_nlp.average = my_metrics.aNlp.average;
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metrics->a_nlp.maximum = my_metrics.aNlp.max;
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metrics->a_nlp.minimum = my_metrics.aNlp.min;
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metrics->divergent_filter_fraction = my_metrics.divergent_filter_fraction;
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return AudioProcessing::kNoError;
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}
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bool EchoCancellationImpl::stream_has_echo() const {
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return stream_has_echo_;
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}
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int EchoCancellationImpl::enable_delay_logging(bool enable) {
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delay_logging_enabled_ = enable;
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return Configure();
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}
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bool EchoCancellationImpl::is_delay_logging_enabled() const {
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return enabled_ && delay_logging_enabled_;
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}
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bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
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return delay_agnostic_enabled_;
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}
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std::string EchoCancellationImpl::GetExperimentsDescription() {
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return refined_adaptive_filter_enabled_ ? "RefinedAdaptiveFilter;" : "";
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}
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bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const {
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return refined_adaptive_filter_enabled_;
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}
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bool EchoCancellationImpl::is_extended_filter_enabled() const {
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return extended_filter_enabled_;
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}
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// TODO(bjornv): How should we handle the multi-channel case?
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int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
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float fraction_poor_delays = 0;
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return GetDelayMetrics(median, std, &fraction_poor_delays);
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}
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int EchoCancellationImpl::GetDelayMetrics(int* median,
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int* std,
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float* fraction_poor_delays) {
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if (median == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (std == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (!enabled_ || !delay_logging_enabled_) {
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return AudioProcessing::kNotEnabledError;
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}
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const int err = WebRtcAec_GetDelayMetrics(cancellers_[0]->state(), median,
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std, fraction_poor_delays);
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if (err != AudioProcessing::kNoError) {
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return MapError(err);
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}
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return AudioProcessing::kNoError;
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}
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struct AecCore* EchoCancellationImpl::aec_core() const {
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if (!enabled_) {
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return NULL;
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}
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return WebRtcAec_aec_core(cancellers_[0]->state());
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}
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void EchoCancellationImpl::Initialize(int sample_rate_hz,
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size_t num_reverse_channels,
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size_t num_output_channels,
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size_t num_proc_channels) {
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stream_properties_.reset(
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new StreamProperties(sample_rate_hz, num_reverse_channels,
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num_output_channels, num_proc_channels));
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if (!enabled_) {
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return;
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}
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const size_t num_cancellers_required =
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NumCancellersRequired(stream_properties_->num_output_channels,
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stream_properties_->num_reverse_channels);
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if (num_cancellers_required > cancellers_.size()) {
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const size_t cancellers_old_size = cancellers_.size();
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cancellers_.resize(num_cancellers_required);
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for (size_t i = cancellers_old_size; i < cancellers_.size(); ++i) {
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cancellers_[i].reset(new Canceller());
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}
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}
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for (auto& canceller : cancellers_) {
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canceller->Initialize(sample_rate_hz);
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}
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Configure();
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}
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int EchoCancellationImpl::GetSystemDelayInSamples() const {
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RTC_DCHECK(enabled_);
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// Report the delay for the first AEC component.
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return WebRtcAec_system_delay(WebRtcAec_aec_core(cancellers_[0]->state()));
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}
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void EchoCancellationImpl::PackRenderAudioBuffer(
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const AudioBuffer* audio,
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size_t num_output_channels,
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size_t num_channels,
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std::vector<float>* packed_buffer) {
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RTC_DCHECK_GE(160, audio->num_frames_per_band());
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RTC_DCHECK_EQ(num_channels, audio->num_channels());
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packed_buffer->clear();
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// The ordering convention must be followed to pass the correct data.
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for (size_t i = 0; i < num_output_channels; i++) {
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for (size_t j = 0; j < audio->num_channels(); j++) {
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// Buffer the samples in the render queue.
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packed_buffer->insert(packed_buffer->end(),
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audio->split_bands_const_f(j)[kBand0To8kHz],
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(audio->split_bands_const_f(j)[kBand0To8kHz] +
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audio->num_frames_per_band()));
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}
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}
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}
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void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
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||
|
{
|
||
|
extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
|
||
|
delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
|
||
|
refined_adaptive_filter_enabled_ =
|
||
|
config.Get<RefinedAdaptiveFilter>().enabled;
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||
|
}
|
||
|
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
|