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1 // Copyright 2015 The Chromium Authors. All rights reserved. | |
2 // Use of this source code is governed by a BSD-style license that can be | |
3 // found in the LICENSE file. | |
4 | |
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ | |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ | |
7 | |
8 #include "base/memory/ref_counted.h" | |
9 #include "base/threading/thread_checker.h" | |
10 #include "content/renderer/media/media_stream_audio_track.h" | |
11 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" | |
12 | |
13 namespace content { | |
14 | |
15 class MediaStreamRemoteAudioSource; | |
16 | |
17 // MediaStreamRemoteAudioTrack is a WebRTC specific implementation of an | |
18 // audio track received from a PeerConnection. | |
19 // TODO(tommi): Chrome shouldn't have to care about remote vs local so | |
20 // we should have a single track implementation that delegates to the | |
21 // sources that do different things depending on the type of source. | |
22 class MediaStreamRemoteAudioTrack : public MediaStreamAudioTrack { | |
23 public: | |
24 explicit MediaStreamRemoteAudioTrack( | |
25 const blink::WebMediaStreamSource& source, bool enabled); | |
26 ~MediaStreamRemoteAudioTrack() override; | |
27 | |
28 void SetEnabled(bool enabled) override; | |
29 void Stop() override; | |
30 | |
31 void AddSink(MediaStreamAudioSink* sink) override; | |
32 void RemoveSink(MediaStreamAudioSink* sink) override; | |
33 media::AudioParameters GetOutputFormat() const override; | |
34 | |
35 webrtc::AudioTrackInterface* GetAudioAdapter() override; | |
36 | |
37 private: | |
38 MediaStreamRemoteAudioSource* source() const; | |
39 | |
40 blink::WebMediaStreamSource source_; | |
41 bool enabled_; | |
42 }; | |
43 | |
44 // Inheriting from ExtraData directly since MediaStreamAudioSource has | |
45 // too much unrelated bloat. | |
46 // TODO(tommi): MediaStreamAudioSource needs refactoring. | |
47 class MediaStreamRemoteAudioSource | |
48 : public blink::WebMediaStreamSource::ExtraData { | |
49 public: | |
50 explicit MediaStreamRemoteAudioSource( | |
51 const scoped_refptr<webrtc::AudioTrackInterface>& track); | |
52 ~MediaStreamRemoteAudioSource() override; | |
53 | |
54 // Controls whether or not the source is included in the main, mixed, audio | |
55 // output from WebRTC as rendered by WebRtcAudioRenderer (media players). | |
56 void SetEnabledForMixing(bool enabled); | |
57 | |
58 // Adds an audio sink for a track belonging to this source. | |
59 // |enabled| is the enabled state of the track and can be updated via | |
60 // a call to SetSinksEnabled. | |
61 void AddSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track, | |
62 bool enabled); | |
63 | |
64 // Removes an audio sink for a track belonging to this source. | |
65 void RemoveSink(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track); | |
66 | |
67 // Turns audio callbacks on/off for all sinks belonging to a track. | |
68 void SetSinksEnabled(MediaStreamAudioTrack* track, bool enabled); | |
69 | |
70 // Removes all sinks belonging to a track. | |
71 void RemoveAll(MediaStreamAudioTrack* track); | |
72 | |
73 webrtc::AudioTrackInterface* GetAudioAdapter(); | |
74 | |
75 private: | |
76 class AudioSink; | |
77 scoped_ptr<AudioSink> sink_; | |
78 const scoped_refptr<webrtc::AudioTrackInterface> track_; | |
79 base::ThreadChecker thread_checker_; | |
80 }; | |
81 | |
82 } // namespace content | |
83 | |
84 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_MEDIA_STREAM_REMOTE_AUDIO_TRACK_H_ | |
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