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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 "remoting/protocol/webrtc_audio_source_adapter.h"
#include <utility>
#include "base/bind.h"
#include "base/logging.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "remoting/proto/audio.pb.h"
#include "remoting/protocol/audio_source.h"
namespace remoting {
namespace protocol {
static const int kChannels = 2;
static const int kBytesPerSample = 2;
// Frame size expected by webrtc::AudioTrackSinkInterface.
static constexpr base::TimeDelta kAudioFrameDuration =
base::TimeDelta::FromMilliseconds(10);
class WebrtcAudioSourceAdapter::Core {
public:
Core();
~Core();
void Start(std::unique_ptr<AudioSource> audio_source);
void Pause(bool pause);
void AddSink(webrtc::AudioTrackSinkInterface* sink);
void RemoveSink(webrtc::AudioTrackSinkInterface* sink);
private:
void OnAudioPacket(std::unique_ptr<AudioPacket> packet);
std::unique_ptr<AudioSource> audio_source_;
bool paused_ = false;
int sampling_rate_ = 0;
// webrtc::AudioTrackSinkInterface expects to get audio in 10ms frames (see
// kAudioFrameDuration). AudioSource may generate AudioPackets for time
// intervals that are not multiple of 10ms. In that case the left-over samples
// are kept in |partial_frame_| until the next AudioPacket is captured by the
// AudioSource.
std::vector<uint8_t> partial_frame_;
base::ObserverList<webrtc::AudioTrackSinkInterface> audio_sinks_;
base::Lock audio_sinks_lock_;
base::ThreadChecker thread_checker_;
};
WebrtcAudioSourceAdapter::Core::Core() {
thread_checker_.DetachFromThread();
}
WebrtcAudioSourceAdapter::Core::~Core() {}
void WebrtcAudioSourceAdapter::Core::Start(
std::unique_ptr<AudioSource> audio_source) {
DCHECK(thread_checker_.CalledOnValidThread());
audio_source_ = std::move(audio_source);
audio_source_->Start(
base::Bind(&Core::OnAudioPacket, base::Unretained(this)));
}
void WebrtcAudioSourceAdapter::Core::Pause(bool pause) {
DCHECK(thread_checker_.CalledOnValidThread());
paused_ = pause;
}
void WebrtcAudioSourceAdapter::Core::AddSink(
webrtc::AudioTrackSinkInterface* sink) {
// Can be called on any thread.
base::AutoLock lock(audio_sinks_lock_);
audio_sinks_.AddObserver(sink);
}
void WebrtcAudioSourceAdapter::Core::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) {
// Can be called on any thread.
base::AutoLock lock(audio_sinks_lock_);
audio_sinks_.RemoveObserver(sink);
}
void WebrtcAudioSourceAdapter::Core::OnAudioPacket(
std::unique_ptr<AudioPacket> packet) {
DCHECK(thread_checker_.CalledOnValidThread());
if (paused_)
return;
DCHECK_EQ(packet->channels(), kChannels);
DCHECK_EQ(packet->bytes_per_sample(), kBytesPerSample);
if (sampling_rate_ != packet->sampling_rate()) {
sampling_rate_ = packet->sampling_rate();
partial_frame_.clear();
}
size_t samples_per_frame =
kAudioFrameDuration * sampling_rate_ / base::TimeDelta::FromSeconds(1);
size_t bytes_per_frame = kBytesPerSample * kChannels * samples_per_frame;
const std::string& data = packet->data(0);
size_t position = 0;
base::AutoLock lock(audio_sinks_lock_);
if (!partial_frame_.empty()) {
size_t bytes_to_append =
std::min(bytes_per_frame - partial_frame_.size(), data.size());
position += bytes_to_append;
partial_frame_.insert(partial_frame_.end(), data.data(),
data.data() + bytes_to_append);
if (partial_frame_.size() < bytes_per_frame) {
// Still don't have full frame.
return;
}
// Here |partial_frame_| always contains a full frame.
DCHECK_EQ(partial_frame_.size(), bytes_per_frame);
for (auto& observer : audio_sinks_) {
observer.OnData(&partial_frame_.front(), kBytesPerSample * 8,
sampling_rate_, kChannels, samples_per_frame);
}
}
while (position + bytes_per_frame <= data.size()) {
for (auto& observer : audio_sinks_) {
observer.OnData(data.data() + position, kBytesPerSample * 8,
sampling_rate_, kChannels, samples_per_frame);
}
position += bytes_per_frame;
}
partial_frame_.assign(data.data() + position, data.data() + data.size());
}
WebrtcAudioSourceAdapter::WebrtcAudioSourceAdapter(
scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner)
: audio_task_runner_(std::move(audio_task_runner)), core_(new Core()) {}
WebrtcAudioSourceAdapter::~WebrtcAudioSourceAdapter() {
audio_task_runner_->DeleteSoon(FROM_HERE, core_.release());
}
void WebrtcAudioSourceAdapter::Start(
std::unique_ptr<AudioSource> audio_source) {
audio_task_runner_->PostTask(
FROM_HERE, base::Bind(&Core::Start, base::Unretained(core_.get()),
base::Passed(&audio_source)));
}
void WebrtcAudioSourceAdapter::Pause(bool pause) {
audio_task_runner_->PostTask(
FROM_HERE,
base::Bind(&Core::Pause, base::Unretained(core_.get()), pause));
}
WebrtcAudioSourceAdapter::SourceState WebrtcAudioSourceAdapter::state() const {
return kLive;
}
bool WebrtcAudioSourceAdapter::remote() const {
return false;
}
void WebrtcAudioSourceAdapter::RegisterAudioObserver(AudioObserver* observer) {}
void WebrtcAudioSourceAdapter::UnregisterAudioObserver(
AudioObserver* observer) {}
void WebrtcAudioSourceAdapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
core_->AddSink(sink);
}
void WebrtcAudioSourceAdapter::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) {
core_->RemoveSink(sink);
}
void WebrtcAudioSourceAdapter::RegisterObserver(
webrtc::ObserverInterface* observer) {}
void WebrtcAudioSourceAdapter::UnregisterObserver(
webrtc::ObserverInterface* observer) {}
} // namespace protocol
} // namespace remoting
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