Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(456)

Side by Side Diff: webrtc/api/remoteaudiosource.h

Issue 2046173002: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moving code that needs to execute out of RTC_DCHECKs. Created 4 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/api/peerconnection.cc ('k') | webrtc/api/remoteaudiosource.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2014 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_ 11 #ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_
12 #define WEBRTC_API_REMOTEAUDIOSOURCE_H_ 12 #define WEBRTC_API_REMOTEAUDIOSOURCE_H_
13 13
14 #include <list> 14 #include <list>
15 #include <string> 15 #include <string>
16 16
17 #include "webrtc/api/mediastreaminterface.h"
18 #include "webrtc/api/notifier.h" 17 #include "webrtc/api/notifier.h"
19 #include "webrtc/audio_sink.h" 18 #include "webrtc/audio_sink.h"
20 #include "webrtc/base/criticalsection.h" 19 #include "webrtc/base/criticalsection.h"
20 #include "webrtc/pc/channel.h"
21 21
22 namespace rtc { 22 namespace rtc {
23 struct Message; 23 struct Message;
24 class Thread; 24 class Thread;
25 } // namespace rtc 25 } // namespace rtc
26 26
27 namespace webrtc { 27 namespace webrtc {
28 28
29 class AudioProviderInterface;
30
31 // This class implements the audio source used by the remote audio track. 29 // This class implements the audio source used by the remote audio track.
32 class RemoteAudioSource : public Notifier<AudioSourceInterface> { 30 class RemoteAudioSource : public Notifier<AudioSourceInterface> {
33 public: 31 public:
34 // Creates an instance of RemoteAudioSource. 32 // Creates an instance of RemoteAudioSource.
35 static rtc::scoped_refptr<RemoteAudioSource> Create( 33 static rtc::scoped_refptr<RemoteAudioSource> Create(
36 uint32_t ssrc, 34 uint32_t ssrc,
37 AudioProviderInterface* provider); 35 cricket::VoiceChannel* channel);
38 36
39 // MediaSourceInterface implementation. 37 // MediaSourceInterface implementation.
40 MediaSourceInterface::SourceState state() const override; 38 MediaSourceInterface::SourceState state() const override;
41 bool remote() const override; 39 bool remote() const override;
42 40
43 void AddSink(AudioTrackSinkInterface* sink) override; 41 void AddSink(AudioTrackSinkInterface* sink) override;
44 void RemoveSink(AudioTrackSinkInterface* sink) override; 42 void RemoveSink(AudioTrackSinkInterface* sink) override;
45 43
46 protected: 44 protected:
47 RemoteAudioSource(); 45 RemoteAudioSource();
48 ~RemoteAudioSource() override; 46 ~RemoteAudioSource() override;
49 47
50 // Post construction initialize where we can do things like save a reference 48 // Post construction initialize where we can do things like save a reference
51 // to ourselves (need to be fully constructed). 49 // to ourselves (need to be fully constructed).
52 void Initialize(uint32_t ssrc, AudioProviderInterface* provider); 50 void Initialize(uint32_t ssrc, cricket::VoiceChannel* channel);
53 51
54 private: 52 private:
55 typedef std::list<AudioObserver*> AudioObserverList; 53 typedef std::list<AudioObserver*> AudioObserverList;
56 54
57 // AudioSourceInterface implementation. 55 // AudioSourceInterface implementation.
58 void SetVolume(double volume) override; 56 void SetVolume(double volume) override;
59 void RegisterAudioObserver(AudioObserver* observer) override; 57 void RegisterAudioObserver(AudioObserver* observer) override;
60 void UnregisterAudioObserver(AudioObserver* observer) override; 58 void UnregisterAudioObserver(AudioObserver* observer) override;
61 59
62 class Sink; 60 class Sink;
63 void OnData(const AudioSinkInterface::Data& audio); 61 void OnData(const AudioSinkInterface::Data& audio);
64 void OnAudioProviderGone(); 62 void OnAudioChannelGone();
65 63
66 class MessageHandler; 64 class MessageHandler;
67 void OnMessage(rtc::Message* msg); 65 void OnMessage(rtc::Message* msg);
68 66
69 AudioObserverList audio_observers_; 67 AudioObserverList audio_observers_;
70 rtc::CriticalSection sink_lock_; 68 rtc::CriticalSection sink_lock_;
71 std::list<AudioTrackSinkInterface*> sinks_; 69 std::list<AudioTrackSinkInterface*> sinks_;
72 rtc::Thread* const main_thread_; 70 rtc::Thread* const main_thread_;
73 SourceState state_; 71 SourceState state_;
74 }; 72 };
75 73
76 } // namespace webrtc 74 } // namespace webrtc
77 75
78 #endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_ 76 #endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_
OLDNEW
« no previous file with comments | « webrtc/api/peerconnection.cc ('k') | webrtc/api/remoteaudiosource.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698