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/*
* libjingle
* Copyright 2014 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/remoteaudiosource.h"
#include <algorithm>
#include <functional>
#include <utility>
#include "talk/app/webrtc/mediastreamprovider.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
namespace webrtc {
class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
public:
explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
private:
~MessageHandler() override {}
void OnMessage(rtc::Message* msg) override {
source_->OnMessage(msg);
delete this;
}
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler);
};
class RemoteAudioSource::Sink : public AudioSinkInterface {
public:
explicit Sink(RemoteAudioSource* source) : source_(source) {}
~Sink() override { source_->OnAudioProviderGone(); }
private:
void OnData(const AudioSinkInterface::Data& audio) override {
if (source_)
source_->OnData(audio);
}
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
};
rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
uint32_t ssrc,
AudioProviderInterface* provider) {
rtc::scoped_refptr<RemoteAudioSource> ret(
new rtc::RefCountedObject<RemoteAudioSource>());
ret->Initialize(ssrc, provider);
return ret;
}
RemoteAudioSource::RemoteAudioSource()
: main_thread_(rtc::Thread::Current()),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(audio_observers_.empty());
RTC_DCHECK(sinks_.empty());
}
void RemoteAudioSource::Initialize(uint32_t ssrc,
AudioProviderInterface* provider) {
RTC_DCHECK(main_thread_->IsCurrent());
// To make sure we always get notified when the provider goes out of scope,
// we register for callbacks here and not on demand in AddSink.
if (provider) { // May be null in tests.
provider->SetRawAudioSink(
ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
}
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK(main_thread_->IsCurrent());
return true;
}
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK(volume >= 0 && volume <= 10);
for (auto* observer : audio_observers_)
observer->OnSetVolume(volume);
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(),
observer) == audio_observers_.end());
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
audio_observers_.remove(observer);
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
if (state_ != MediaSourceInterface::kLive) {
LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
return;
}
rtc::CritScope lock(&sink_lock_);
RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
rtc::CritScope lock(&sink_lock_);
sinks_.remove(sink);
}
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
rtc::CritScope lock(&sink_lock_);
for (auto* sink : sinks_) {
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel);
}
}
void RemoteAudioSource::OnAudioProviderGone() {
// Called when the data provider is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
main_thread_->Post(new MessageHandler(this));
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
state_ = MediaSourceInterface::kEnded;
FireOnChanged();
}
} // namespace webrtc
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