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// Copyright 2013 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/mediastream/media_stream_audio_processor.h"
#include <memory>
#include <optional>
#include <string_view>
#include "base/memory/raw_ptr.h"
#include "base/task/single_thread_task_runner.h"
#include "build/build_config.h"
#include "media/base/audio_parameters.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h"
#include "third_party/blink/renderer/platform/mediastream/aec_dump_agent_impl.h"
#include "third_party/blink/renderer/platform/wtf/functional.h"
namespace blink {
namespace {
void WebRtcLogStringPiece(std::string_view message) {
WebRtcLogMessage(std::string{message});
}
} // namespace
// Subscribes a sink to the playout data source for the duration of the
// PlayoutListener lifetime.
class MediaStreamAudioProcessor::PlayoutListener {
public:
PlayoutListener(scoped_refptr<WebRtcAudioDeviceImpl> playout_data_source,
WebRtcPlayoutDataSource::Sink* sink)
: playout_data_source_(std::move(playout_data_source)), sink_(sink) {
DCHECK(playout_data_source_);
DCHECK(sink_);
playout_data_source_->AddPlayoutSink(sink_);
}
~PlayoutListener() { playout_data_source_->RemovePlayoutSink(sink_); }
private:
// TODO(crbug.com/704136): Replace with Member at some point.
scoped_refptr<WebRtcAudioDeviceImpl> const playout_data_source_;
const raw_ptr<WebRtcPlayoutDataSource::Sink> sink_;
};
MediaStreamAudioProcessor::MediaStreamAudioProcessor(
DeliverProcessedAudioCallback deliver_processed_audio_callback,
const media::AudioProcessingSettings& settings,
const media::AudioParameters& capture_data_source_params,
scoped_refptr<WebRtcAudioDeviceImpl> playout_data_source)
: audio_processor_(media::AudioProcessor::Create(
std::move(deliver_processed_audio_callback),
/*log_callback=*/
WTF::BindRepeating(&WebRtcLogStringPiece),
settings,
capture_data_source_params,
media::AudioProcessor::GetDefaultOutputFormat(
capture_data_source_params,
settings))),
main_thread_runner_(base::SingleThreadTaskRunner::GetCurrentDefault()),
aec_dump_agent_impl_(AecDumpAgentImpl::Create(this)),
stopped_(false) {
DCHECK(main_thread_runner_);
// Register as a listener for the playout reference signal. Used for e.g. echo
// cancellation.
if (audio_processor_->needs_playout_reference() && playout_data_source) {
playout_listener_ =
std::make_unique<PlayoutListener>(std::move(playout_data_source), this);
}
DETACH_FROM_THREAD(capture_thread_checker_);
DETACH_FROM_THREAD(render_thread_checker_);
}
MediaStreamAudioProcessor::~MediaStreamAudioProcessor() {
DCHECK(main_thread_runner_->BelongsToCurrentThread());
Stop();
}
void MediaStreamAudioProcessor::ProcessCapturedAudio(
const media::AudioBus& audio_source,
base::TimeTicks audio_capture_time,
int num_preferred_channels,
double volume) {
DCHECK_CALLED_ON_VALID_THREAD(capture_thread_checker_);
audio_processor_->ProcessCapturedAudio(audio_source, audio_capture_time,
num_preferred_channels, volume);
}
void MediaStreamAudioProcessor::Stop() {
DCHECK(main_thread_runner_->BelongsToCurrentThread());
if (stopped_)
return;
stopped_ = true;
aec_dump_agent_impl_.reset();
audio_processor_->OnStopDump();
playout_listener_.reset();
}
const media::AudioParameters&
MediaStreamAudioProcessor::GetInputFormatForTesting() const {
return audio_processor_->input_format();
}
void MediaStreamAudioProcessor::OnStartDump(base::File dump_file) {
DCHECK(main_thread_runner_->BelongsToCurrentThread());
audio_processor_->OnStartDump(std::move(dump_file));
}
void MediaStreamAudioProcessor::OnStopDump() {
DCHECK(main_thread_runner_->BelongsToCurrentThread());
audio_processor_->OnStopDump();
}
// static
// TODO(https://crbug.com/1269364): This logic should be moved to
// ProcessedLocalAudioSource and verified/fixed; The decision should be
// "hardware effects are required or software audio mofidications are needed
// (AudioProcessingSettings.NeedAudioModification())".
bool MediaStreamAudioProcessor::WouldModifyAudio(
const AudioProcessingProperties& properties) {
if (properties
.ToAudioProcessingSettings(
/*multi_channel_capture_processing - does not matter here*/ false)
.NeedWebrtcAudioProcessing()) {
return true;
}
#if !BUILDFLAG(IS_IOS)
if (properties.auto_gain_control) {
return true;
}
#endif
if (properties.noise_suppression) {
return true;
}
return false;
}
void MediaStreamAudioProcessor::OnPlayoutData(media::AudioBus* audio_bus,
int sample_rate,
base::TimeDelta audio_delay) {
DCHECK_CALLED_ON_VALID_THREAD(render_thread_checker_);
DCHECK(audio_bus);
audio_processor_->OnPlayoutData(*audio_bus, sample_rate, audio_delay);
}
void MediaStreamAudioProcessor::OnPlayoutDataSourceChanged() {
DCHECK(main_thread_runner_->BelongsToCurrentThread());
DETACH_FROM_THREAD(render_thread_checker_);
}
void MediaStreamAudioProcessor::OnRenderThreadChanged() {
DETACH_FROM_THREAD(render_thread_checker_);
DCHECK_CALLED_ON_VALID_THREAD(render_thread_checker_);
}
webrtc::AudioProcessorInterface::AudioProcessorStatistics
MediaStreamAudioProcessor::GetStats(bool has_remote_tracks) {
AudioProcessorStatistics stats;
stats.apm_statistics = audio_processor_->GetStats();
return stats;
}
} // namespace blink
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