/*
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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 "webrtc/modules/audio_processing/echo_cancellation_impl.h"
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#include <assert.h>
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#include <string.h>
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extern "C" {
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#include "webrtc/modules/audio_processing/aec/aec_core.h"
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}
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#include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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namespace webrtc {
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typedef void Handle;
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namespace {
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int16_t MapSetting(EchoCancellation::SuppressionLevel level) {
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switch (level) {
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case EchoCancellation::kLowSuppression:
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return kAecNlpConservative;
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case EchoCancellation::kModerateSuppression:
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return kAecNlpModerate;
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case EchoCancellation::kHighSuppression:
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return kAecNlpAggressive;
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}
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assert(false);
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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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// Maximum length that a frame of samples can have.
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static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
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// Maximum number of frames to buffer in the render queue.
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// TODO(peah): Decrease this once we properly handle hugely unbalanced
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// reverse and forward call numbers.
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static const size_t kMaxNumFramesToBuffer = 100;
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} // namespace
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EchoCancellationImpl::EchoCancellationImpl(const AudioProcessing* apm,
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rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture)
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: ProcessingComponent(),
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apm_(apm),
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crit_render_(crit_render),
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crit_capture_(crit_capture),
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drift_compensation_enabled_(false),
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metrics_enabled_(false),
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suppression_level_(kModerateSuppression),
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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_(false),
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extended_filter_enabled_(false),
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delay_agnostic_enabled_(false),
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render_queue_element_max_size_(0) {
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RTC_DCHECK(apm);
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RTC_DCHECK(crit_render);
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RTC_DCHECK(crit_capture);
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}
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EchoCancellationImpl::~EchoCancellationImpl() {}
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int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
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rtc::CritScope cs_render(crit_render_);
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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assert(audio->num_frames_per_band() <= 160);
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assert(audio->num_channels() == apm_->num_reverse_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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render_queue_buffer_.clear();
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for (size_t i = 0; i < apm_->num_output_channels(); i++) {
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for (size_t j = 0; j < audio->num_channels(); j++) {
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Handle* my_handle = static_cast<Handle*>(handle(handle_index));
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// Retrieve any error code produced by the buffering of the farend
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// signal
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err = WebRtcAec_GetBufferFarendError(
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my_handle, audio->split_bands_const_f(j)[kBand0To8kHz],
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audio->num_frames_per_band());
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if (err != AudioProcessing::kNoError) {
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return MapError(err); // TODO(ajm): warning possible?
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}
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// Buffer the samples in the render queue.
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render_queue_buffer_.insert(render_queue_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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// Insert the samples into the queue.
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if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
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// The data queue is full and needs to be emptied.
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ReadQueuedRenderData();
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// Retry the insert (should always work).
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RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
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}
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return AudioProcessing::kNoError;
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}
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// Read chunks of data that were received and queued on the render side from
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// a queue. All the data chunks are buffered into the farend signal of the AEC.
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void EchoCancellationImpl::ReadQueuedRenderData() {
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rtc::CritScope cs_capture(crit_capture_);
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if (!is_component_enabled()) {
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return;
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}
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while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
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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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capture_queue_buffer_.size() /
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(apm_->num_output_channels() * apm_->num_reverse_channels());
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for (size_t i = 0; i < apm_->num_output_channels(); i++) {
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for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
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Handle* my_handle = static_cast<Handle*>(handle(handle_index));
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WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
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num_frames_per_band);
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buffer_index += num_frames_per_band;
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handle_index++;
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}
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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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rtc::CritScope cs_capture(crit_capture_);
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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if (!apm_->was_stream_delay_set()) {
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return AudioProcessing::kStreamParameterNotSetError;
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}
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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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assert(audio->num_frames_per_band() <= 160);
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assert(audio->num_channels() == apm_->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 < apm_->num_reverse_channels(); j++) {
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Handle* my_handle = handle(handle_index);
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err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
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audio->num_bands(), audio->split_bands_f(i),
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audio->num_frames_per_band(),
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apm_->stream_delay_ms(), 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(my_handle, &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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// Run in a single-threaded manner.
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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// Ensure AEC and AECM are not both enabled.
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// The is_enabled call is safe from a deadlock perspective
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// as both locks are already held in the correct order.
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if (enable && apm_->echo_control_mobile()->is_enabled()) {
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return AudioProcessing::kBadParameterError;
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}
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return EnableComponent(enable);
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}
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bool EchoCancellationImpl::is_enabled() const {
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rtc::CritScope cs(crit_capture_);
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return is_component_enabled();
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}
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int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
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{
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if (MapSetting(level) == -1) {
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return AudioProcessing::kBadParameterError;
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}
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rtc::CritScope cs(crit_capture_);
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suppression_level_ = level;
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}
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return Configure();
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}
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EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
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const {
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rtc::CritScope cs(crit_capture_);
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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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{
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rtc::CritScope cs(crit_capture_);
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drift_compensation_enabled_ = enable;
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}
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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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rtc::CritScope cs(crit_capture_);
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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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rtc::CritScope cs(crit_capture_);
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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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rtc::CritScope cs(crit_capture_);
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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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{
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rtc::CritScope cs(crit_capture_);
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metrics_enabled_ = enable;
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}
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return Configure();
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}
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bool EchoCancellationImpl::are_metrics_enabled() const {
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rtc::CritScope cs(crit_capture_);
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return 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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rtc::CritScope cs(crit_capture_);
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if (metrics == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (!is_component_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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Handle* my_handle = static_cast<Handle*>(handle(0));
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int err = WebRtcAec_GetMetrics(my_handle, &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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return AudioProcessing::kNoError;
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}
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bool EchoCancellationImpl::stream_has_echo() const {
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rtc::CritScope cs(crit_capture_);
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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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{
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rtc::CritScope cs(crit_capture_);
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delay_logging_enabled_ = enable;
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}
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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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rtc::CritScope cs(crit_capture_);
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return delay_logging_enabled_;
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}
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bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
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rtc::CritScope cs(crit_capture_);
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return delay_agnostic_enabled_;
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}
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bool EchoCancellationImpl::is_extended_filter_enabled() const {
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rtc::CritScope cs(crit_capture_);
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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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rtc::CritScope cs(crit_capture_);
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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, int* std,
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float* fraction_poor_delays) {
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rtc::CritScope cs(crit_capture_);
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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 (!is_component_enabled() || !delay_logging_enabled_) {
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return AudioProcessing::kNotEnabledError;
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}
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Handle* my_handle = static_cast<Handle*>(handle(0));
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const int err =
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WebRtcAec_GetDelayMetrics(my_handle, median, 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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rtc::CritScope cs(crit_capture_);
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if (!is_component_enabled()) {
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return NULL;
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}
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Handle* my_handle = static_cast<Handle*>(handle(0));
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return WebRtcAec_aec_core(my_handle);
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}
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int EchoCancellationImpl::Initialize() {
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int err = ProcessingComponent::Initialize();
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{
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rtc::CritScope cs(crit_capture_);
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if (err != AudioProcessing::kNoError || !is_component_enabled()) {
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return err;
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}
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}
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AllocateRenderQueue();
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return AudioProcessing::kNoError;
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}
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void EchoCancellationImpl::AllocateRenderQueue() {
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const size_t new_render_queue_element_max_size = std::max<size_t>(
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static_cast<size_t>(1),
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kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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// Reallocate the queue if the queue item size is too small to fit the
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// data to put in the queue.
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if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
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render_queue_element_max_size_ = new_render_queue_element_max_size;
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std::vector<float> template_queue_element(render_queue_element_max_size_);
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render_signal_queue_.reset(
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new SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>(
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kMaxNumFramesToBuffer, template_queue_element,
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RenderQueueItemVerifier<float>(render_queue_element_max_size_)));
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render_queue_buffer_.resize(render_queue_element_max_size_);
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capture_queue_buffer_.resize(render_queue_element_max_size_);
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} else {
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render_signal_queue_->Clear();
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}
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}
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void EchoCancellationImpl::SetExtraOptions(const Config& config) {
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{
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rtc::CritScope cs(crit_capture_);
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extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
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delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
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}
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Configure();
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}
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void* EchoCancellationImpl::CreateHandle() const {
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return WebRtcAec_Create();
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}
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void EchoCancellationImpl::DestroyHandle(void* handle) const {
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assert(handle != NULL);
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WebRtcAec_Free(static_cast<Handle*>(handle));
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}
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int EchoCancellationImpl::InitializeHandle(void* handle) const {
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// Not locked as it only relies on APM public API which is threadsafe.
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assert(handle != NULL);
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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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return WebRtcAec_Init(static_cast<Handle*>(handle),
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apm_->proc_sample_rate_hz(), 48000);
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}
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int EchoCancellationImpl::ConfigureHandle(void* handle) const {
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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assert(handle != NULL);
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AecConfig config;
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config.metricsMode = metrics_enabled_;
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config.nlpMode = MapSetting(suppression_level_);
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config.skewMode = drift_compensation_enabled_;
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config.delay_logging = delay_logging_enabled_;
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WebRtcAec_enable_extended_filter(
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WebRtcAec_aec_core(static_cast<Handle*>(handle)),
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extended_filter_enabled_ ? 1 : 0);
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WebRtcAec_enable_delay_agnostic(
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WebRtcAec_aec_core(static_cast<Handle*>(handle)),
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delay_agnostic_enabled_ ? 1 : 0);
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return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
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}
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size_t EchoCancellationImpl::num_handles_required() const {
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// Not locked as it only relies on APM public API which is threadsafe.
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return apm_->num_output_channels() * apm_->num_reverse_channels();
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}
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int EchoCancellationImpl::GetHandleError(void* handle) const {
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// Not locked as it does not rely on anything in the state.
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assert(handle != NULL);
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return AudioProcessing::kUnspecifiedError;
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}
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} // namespace webrtc
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