7ffeab525c
This is a reland of https://codereview.webrtc.org/1737593002/ minus the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and webrtc/common.gyp that breaks GN in Chromium since it's using the --check flag (which we should support). BUG=webrtc:4243, webrtc:5589 TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully. TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1740873003 . Cr-Commit-Position: refs/heads/master@{#11794}
80 lines
2.3 KiB
C++
80 lines
2.3 KiB
C++
/*
|
|
* Copyright 2014 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
|
|
#define WEBRTC_API_REMOTEAUDIOSOURCE_H_
|
|
|
|
#include <list>
|
|
#include <string>
|
|
|
|
#include "webrtc/api/mediastreaminterface.h"
|
|
#include "webrtc/api/notifier.h"
|
|
#include "webrtc/audio_sink.h"
|
|
#include "webrtc/base/criticalsection.h"
|
|
#include "webrtc/media/base/audiorenderer.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 // WEBRTC_API_REMOTEAUDIOSOURCE_H_
|