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wu@webrtc.orgb9a088b2014-02-13 23:18:49 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2014 Google Inc.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +00004 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
29#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
30
31#include <list>
Tommif888bb52015-12-12 01:37:01 +010032#include <string>
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000033
34#include "talk/app/webrtc/mediastreaminterface.h"
35#include "talk/app/webrtc/notifier.h"
Tommif888bb52015-12-12 01:37:01 +010036#include "talk/media/base/audiorenderer.h"
37#include "webrtc/audio/audio_sink.h"
38#include "webrtc/base/criticalsection.h"
39
40namespace rtc {
41struct Message;
42class Thread;
43} // namespace rtc
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000044
45namespace webrtc {
46
Tommif888bb52015-12-12 01:37:01 +010047class AudioProviderInterface;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000048
49// This class implements the audio source used by the remote audio track.
50class RemoteAudioSource : public Notifier<AudioSourceInterface> {
51 public:
52 // Creates an instance of RemoteAudioSource.
Tommif888bb52015-12-12 01:37:01 +010053 static rtc::scoped_refptr<RemoteAudioSource> Create(
54 uint32_t ssrc,
55 AudioProviderInterface* provider);
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000056
57 // MediaSourceInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000058 MediaSourceInterface::SourceState state() const override;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000059
Tommif888bb52015-12-12 01:37:01 +010060 void AddSink(AudioTrackSinkInterface* sink);
61 void RemoveSink(AudioTrackSinkInterface* sink);
62
63 protected:
64 RemoteAudioSource();
65 ~RemoteAudioSource() override;
66
67 // Post construction initialize where we can do things like save a reference
68 // to ourselves (need to be fully constructed).
69 void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
70
71 private:
72 typedef std::list<AudioObserver*> AudioObserverList;
73
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000074 // AudioSourceInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000075 void SetVolume(double volume) override;
76 void RegisterAudioObserver(AudioObserver* observer) override;
77 void UnregisterAudioObserver(AudioObserver* observer) override;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000078
Tommif888bb52015-12-12 01:37:01 +010079 class Sink;
80 void OnData(const AudioSinkInterface::Data& audio);
81 void OnAudioProviderGone();
82
83 class MessageHandler;
84 void OnMessage(rtc::Message* msg);
85
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000086 AudioObserverList audio_observers_;
Tommif888bb52015-12-12 01:37:01 +010087 rtc::CriticalSection sink_lock_;
88 std::list<AudioTrackSinkInterface*> sinks_;
89 rtc::Thread* const main_thread_;
90 SourceState state_;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +000091};
92
93} // namespace webrtc
94
95#endif // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_