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Side by Side Diff: talk/app/webrtc/remoteaudiosource.h

Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Change when we fire callbacks for external media Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 11 matching lines...) Expand all
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 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 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ 28 #ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
29 #define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ 29 #define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
30 30
31 #include <list> 31 #include <list>
32 #include <string>
32 33
33 #include "talk/app/webrtc/mediastreaminterface.h" 34 #include "talk/app/webrtc/mediastreaminterface.h"
34 #include "talk/app/webrtc/notifier.h" 35 #include "talk/app/webrtc/notifier.h"
36 #include "talk/media/base/audiorenderer.h"
37 #include "webrtc/base/criticalsection.h"
38
39 namespace rtc {
40 struct Message;
41 class Thread;
42 } // namespace rtc
35 43
36 namespace webrtc { 44 namespace webrtc {
37 45
38 using webrtc::AudioSourceInterface; 46 class AudioProviderInterface;
39 47
40 // This class implements the audio source used by the remote audio track. 48 // This class implements the audio source used by the remote audio track.
41 class RemoteAudioSource : public Notifier<AudioSourceInterface> { 49 class RemoteAudioSource : public Notifier<AudioSourceInterface>,
50 public cricket::AudioRenderer::Sink {
42 public: 51 public:
43 // Creates an instance of RemoteAudioSource. 52 // Creates an instance of RemoteAudioSource.
44 static rtc::scoped_refptr<RemoteAudioSource> Create(); 53 static rtc::scoped_refptr<RemoteAudioSource> Create(
54 uint32_t ssrc,
55 AudioProviderInterface* provider);
56
57 void AddSink(AudioTrackSinkInterface* sink);
58 void RemoveSink(AudioTrackSinkInterface* sink);
45 59
46 protected: 60 protected:
47 RemoteAudioSource(); 61 RemoteAudioSource(uint32_t ssrc, AudioProviderInterface* provider);
48 virtual ~RemoteAudioSource(); 62 ~RemoteAudioSource() override;
49 63
50 private: 64 private:
51 typedef std::list<AudioObserver*> AudioObserverList; 65 typedef std::list<AudioObserver*> AudioObserverList;
52 66
53 // MediaSourceInterface implementation. 67 // MediaSourceInterface implementation.
54 MediaSourceInterface::SourceState state() const override; 68 MediaSourceInterface::SourceState state() const override;
55 69
56 // AudioSourceInterface implementation. 70 // AudioSourceInterface implementation.
57 void SetVolume(double volume) override; 71 void SetVolume(double volume) override;
58 void RegisterAudioObserver(AudioObserver* observer) override; 72 void RegisterAudioObserver(AudioObserver* observer) override;
59 void UnregisterAudioObserver(AudioObserver* observer) override; 73 void UnregisterAudioObserver(AudioObserver* observer) override;
60 74
75 // cricket::AudioRenderer::Sink implementation.
76 void OnData(const void* audio_data,
77 int bits_per_sample,
78 int sample_rate,
79 int number_of_channels,
80 size_t number_of_frames) override;
81 void OnClose() override;
82
83 class MessageHandler;
84 void OnMessage(rtc::Message* msg);
85
61 AudioObserverList audio_observers_; 86 AudioObserverList audio_observers_;
87 const uint32_t ssrc_;
88 AudioProviderInterface* provider_;
89 rtc::CriticalSection sink_lock_;
90 std::list<AudioTrackSinkInterface*> sinks_;
91 rtc::Thread* const main_thread_;
92 SourceState state_;
62 }; 93 };
63 94
64 } // namespace webrtc 95 } // namespace webrtc
65 96
66 #endif // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ 97 #endif // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
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