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// Copyright 2016 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "content/renderer/media/webrtc/webrtc_audio_sink.h"
#include <algorithm>
#include <limits>
#include "base/bind.h"
#include "base/bind_helpers.h"
#include "base/location.h"
#include "base/logging.h"
#include "base/threading/thread_task_runner_handle.h"
namespace content {
WebRtcAudioSink::WebRtcAudioSink(
const std::string& label,
scoped_refptr<webrtc::AudioSourceInterface> track_source,
scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
: adapter_(new rtc::RefCountedObject<Adapter>(
label, std::move(track_source), std::move(signaling_task_runner))),
fifo_(base::Bind(&WebRtcAudioSink::DeliverRebufferedAudio,
base::Unretained(this))) {
DVLOG(1) << "WebRtcAudioSink::WebRtcAudioSink()";
}
WebRtcAudioSink::~WebRtcAudioSink() {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebRtcAudioSink::~WebRtcAudioSink()";
}
void WebRtcAudioSink::SetAudioProcessor(
scoped_refptr<MediaStreamAudioProcessor> processor) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(processor.get());
adapter_->set_processor(std::move(processor));
}
void WebRtcAudioSink::SetLevel(
scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(level.get());
adapter_->set_level(std::move(level));
}
void WebRtcAudioSink::OnEnabledChanged(bool enabled) {
DCHECK(thread_checker_.CalledOnValidThread());
adapter_->signaling_task_runner()->PostTask(
FROM_HERE,
base::Bind(
base::IgnoreResult(&WebRtcAudioSink::Adapter::set_enabled),
adapter_, enabled));
}
void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus,
base::TimeTicks estimated_capture_time) {
DCHECK(audio_thread_checker_.CalledOnValidThread());
// The following will result in zero, one, or multiple synchronous calls to
// DeliverRebufferedAudio().
fifo_.Push(audio_bus);
}
void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) {
// On a format change, the thread delivering audio might have also changed.
audio_thread_checker_.DetachFromThread();
DCHECK(audio_thread_checker_.CalledOnValidThread());
DCHECK(params.IsValid());
params_ = params;
// Make sure that our params always reflect a buffer size of 10ms.
params_.set_frames_per_buffer(params_.sample_rate() / 100);
fifo_.Reset(params_.frames_per_buffer());
const int num_pcm16_data_elements =
params_.frames_per_buffer() * params_.channels();
interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);
}
void WebRtcAudioSink::DeliverRebufferedAudio(const media::AudioBus& audio_bus,
int frame_delay) {
DCHECK(audio_thread_checker_.CalledOnValidThread());
DCHECK(params_.IsValid());
// TODO(miu): Why doesn't a WebRTC sink care about reference time passed to
// OnData(), and the |frame_delay| here? How is AV sync achieved otherwise?
// TODO(henrika): Remove this conversion once the interface in libjingle
// supports float vectors.
audio_bus.ToInterleaved(audio_bus.frames(),
sizeof(interleaved_data_[0]),
interleaved_data_.get());
adapter_->DeliverPCMToWebRtcSinks(interleaved_data_.get(),
params_.sample_rate(),
audio_bus.channels(),
audio_bus.frames());
}
namespace {
// TODO(miu): MediaStreamAudioProcessor destructor requires this nonsense.
void DereferenceOnMainThread(
const scoped_refptr<MediaStreamAudioProcessor>& processor) {}
} // namespace
WebRtcAudioSink::Adapter::Adapter(
const std::string& label,
scoped_refptr<webrtc::AudioSourceInterface> source,
scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
: webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
source_(std::move(source)),
signaling_task_runner_(std::move(signaling_task_runner)),
main_task_runner_(base::ThreadTaskRunnerHandle::Get()) {
DCHECK(signaling_task_runner_);
}
WebRtcAudioSink::Adapter::~Adapter() {
if (audio_processor_) {
main_task_runner_->PostTask(
FROM_HERE,
base::Bind(&DereferenceOnMainThread, std::move(audio_processor_)));
}
}
void WebRtcAudioSink::Adapter::DeliverPCMToWebRtcSinks(
const int16_t* audio_data,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {
base::AutoLock auto_lock(lock_);
for (webrtc::AudioTrackSinkInterface* sink : sinks_) {
sink->OnData(audio_data, sizeof(int16_t) * 8, sample_rate,
number_of_channels, number_of_frames);
}
}
std::string WebRtcAudioSink::Adapter::kind() const {
return webrtc::MediaStreamTrackInterface::kAudioKind;
}
bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
set_enabled(enable);
}
void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
DCHECK(sink);
base::AutoLock auto_lock(lock_);
DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
sinks_.push_back(sink);
}
void WebRtcAudioSink::Adapter::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
base::AutoLock auto_lock(lock_);
const auto it = std::find(sinks_.begin(), sinks_.end(), sink);
if (it != sinks_.end())
sinks_.erase(it);
}
bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
// |level_| is only set once, so it's safe to read without first acquiring a
// mutex.
if (!level_)
return false;
const float signal_level = level_->GetCurrent();
DCHECK_GE(signal_level, 0.0f);
DCHECK_LE(signal_level, 1.0f);
// Convert from float in range [0.0,1.0] to an int in range [0,32767].
*level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
0.5f /* rounding to nearest int */);
return true;
}
rtc::scoped_refptr<webrtc::AudioProcessorInterface>
WebRtcAudioSink::Adapter::GetAudioProcessor() {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
return audio_processor_.get();
}
webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksOnCurrentThread());
return source_.get();
}
} // namespace content
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