6eca7e3c37
Additionally: * Moving all implementation inside RemoteAudioTrack into AudioTrack and remove RemoteAudioTrack. * AddSink/RemoveSink are now on all audio sources (like they are for video sources). While doing this I found that some of our tests are broken :) and fixed them. They were broken because AudioTrack didn't previously do much such as updating its state. BUG=chromium:569526 Review URL: https://codereview.webrtc.org/1522903002 Cr-Commit-Position: refs/heads/master@{#11026}
97 lines
3.4 KiB
C++
97 lines
3.4 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
|
|
#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
|
|
|
|
#include <list>
|
|
#include <string>
|
|
|
|
#include "talk/app/webrtc/mediastreaminterface.h"
|
|
#include "talk/app/webrtc/notifier.h"
|
|
#include "talk/media/base/audiorenderer.h"
|
|
#include "webrtc/audio/audio_sink.h"
|
|
#include "webrtc/base/criticalsection.h"
|
|
|
|
namespace rtc {
|
|
struct Message;
|
|
class Thread;
|
|
} // namespace rtc
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioProviderInterface;
|
|
|
|
// This class implements the audio source used by the remote audio track.
|
|
class RemoteAudioSource : public Notifier<AudioSourceInterface> {
|
|
public:
|
|
// Creates an instance of RemoteAudioSource.
|
|
static rtc::scoped_refptr<RemoteAudioSource> Create(
|
|
uint32_t ssrc,
|
|
AudioProviderInterface* provider);
|
|
|
|
// MediaSourceInterface implementation.
|
|
MediaSourceInterface::SourceState state() const override;
|
|
bool remote() const override;
|
|
|
|
void AddSink(AudioTrackSinkInterface* sink) override;
|
|
void RemoveSink(AudioTrackSinkInterface* sink) override;
|
|
|
|
protected:
|
|
RemoteAudioSource();
|
|
~RemoteAudioSource() override;
|
|
|
|
// Post construction initialize where we can do things like save a reference
|
|
// to ourselves (need to be fully constructed).
|
|
void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
|
|
|
|
private:
|
|
typedef std::list<AudioObserver*> AudioObserverList;
|
|
|
|
// AudioSourceInterface implementation.
|
|
void SetVolume(double volume) override;
|
|
void RegisterAudioObserver(AudioObserver* observer) override;
|
|
void UnregisterAudioObserver(AudioObserver* observer) override;
|
|
|
|
class Sink;
|
|
void OnData(const AudioSinkInterface::Data& audio);
|
|
void OnAudioProviderGone();
|
|
|
|
class MessageHandler;
|
|
void OnMessage(rtc::Message* msg);
|
|
|
|
AudioObserverList audio_observers_;
|
|
rtc::CriticalSection sink_lock_;
|
|
std::list<AudioTrackSinkInterface*> sinks_;
|
|
rtc::Thread* const main_thread_;
|
|
SourceState state_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
|