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| 1 // Copyright 2013 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_LOCAL_AUDIO_TRACK_H_ | |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | |
| 7 | |
| 8 #include <list> | |
| 9 #include <string> | |
| 10 | |
| 11 #include "base/macros.h" | |
| 12 #include "base/memory/ref_counted.h" | |
| 13 #include "base/synchronization/lock.h" | |
| 14 #include "base/threading/thread_checker.h" | |
| 15 #include "content/renderer/media/media_stream_audio_track.h" | |
| 16 #include "content/renderer/media/tagged_list.h" | |
| 17 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
| 18 #include "media/audio/audio_parameters.h" | |
| 19 | |
| 20 namespace media { | |
| 21 class AudioBus; | |
| 22 } | |
| 23 | |
| 24 namespace content { | |
| 25 | |
| 26 class MediaStreamAudioLevelCalculator; | |
| 27 class MediaStreamAudioProcessor; | |
| 28 class MediaStreamAudioSink; | |
| 29 class MediaStreamAudioSinkOwner; | |
| 30 class MediaStreamAudioTrackSink; | |
| 31 | |
| 32 // A WebRtcLocalAudioTrack manages thread-safe connects/disconnects to sinks, | |
| 33 // and the delivery of audio data from the source to the sinks. | |
| 34 class CONTENT_EXPORT WebRtcLocalAudioTrack | |
| 35 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) { | |
| 36 public: | |
| 37 explicit WebRtcLocalAudioTrack( | |
| 38 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter); | |
| 39 | |
| 40 ~WebRtcLocalAudioTrack() override; | |
| 41 | |
| 42 // Add a sink to the track. This function will trigger a OnSetFormat() | |
| 43 // call on the |sink|. | |
| 44 // Called on the main render thread. | |
| 45 void AddSink(MediaStreamAudioSink* sink) override; | |
| 46 | |
| 47 // Remove a sink from the track. | |
| 48 // Called on the main render thread. | |
| 49 void RemoveSink(MediaStreamAudioSink* sink) override; | |
| 50 | |
| 51 // Overrides for MediaStreamTrack. | |
| 52 void SetEnabled(bool enabled) override; | |
| 53 webrtc::AudioTrackInterface* GetAudioAdapter() override; | |
| 54 media::AudioParameters GetOutputFormat() const override; | |
| 55 | |
| 56 // Method called by the capturer to deliver the capture data. | |
| 57 // Called on the capture audio thread. | |
| 58 void Capture(const media::AudioBus& audio_bus, | |
| 59 base::TimeTicks estimated_capture_time); | |
| 60 | |
| 61 // Method called by the capturer to set the audio parameters used by source | |
| 62 // of the capture data.. | |
| 63 // Called on the capture audio thread. | |
| 64 void OnSetFormat(const media::AudioParameters& params); | |
| 65 | |
| 66 // Called by the capturer before the audio data flow begins to set the object | |
| 67 // that provides shared access to the current audio signal level. | |
| 68 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level); | |
| 69 | |
| 70 // Called by the capturer before the audio data flow begins to provide a | |
| 71 // reference to the audio processor so that the track can query stats from it. | |
| 72 void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor); | |
| 73 | |
| 74 private: | |
| 75 typedef TaggedList<MediaStreamAudioTrackSink> SinkList; | |
| 76 | |
| 77 // MediaStreamAudioTrack override. | |
| 78 void OnStop() final; | |
| 79 | |
| 80 // All usage of libjingle is through this adapter. The adapter holds | |
| 81 // a pointer to this object, but no reference. | |
| 82 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; | |
| 83 | |
| 84 // A tagged list of sinks that the audio data is fed to. Tags | |
| 85 // indicate tracks that need to be notified that the audio format | |
| 86 // has changed. | |
| 87 SinkList sinks_; | |
| 88 | |
| 89 // Tests that methods are called on libjingle's signaling thread. | |
| 90 base::ThreadChecker signal_thread_checker_; | |
| 91 | |
| 92 // Used to DCHECK that some methods are called on the capture audio thread. | |
| 93 base::ThreadChecker capture_thread_checker_; | |
| 94 | |
| 95 // Protects |params_| and |sinks_|. | |
| 96 mutable base::Lock lock_; | |
| 97 | |
| 98 // Audio parameters of the audio capture stream. | |
| 99 media::AudioParameters audio_parameters_; | |
| 100 | |
| 101 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | |
| 102 }; | |
| 103 | |
| 104 } // namespace content | |
| 105 | |
| 106 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | |
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