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| 1 // Copyright 2014 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_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | |
| 7 | |
| 8 #include <vector> | |
| 9 | |
| 10 #include "base/memory/ref_counted.h" | |
| 11 #include "base/memory/scoped_vector.h" | |
| 12 #include "base/single_thread_task_runner.h" | |
| 13 #include "base/synchronization/lock.h" | |
| 14 #include "content/common/content_export.h" | |
| 15 #include "content/renderer/media/media_stream_audio_level_calculator.h" | |
| 16 #include "content/renderer/media/media_stream_audio_processor.h" | |
| 17 #include "third_party/webrtc/api/mediastreamtrack.h" | |
| 18 #include "third_party/webrtc/media/base/audiorenderer.h" | |
| 19 | |
| 20 namespace cricket { | |
| 21 class AudioRenderer; | |
| 22 } | |
| 23 | |
| 24 namespace webrtc { | |
| 25 class AudioSourceInterface; | |
| 26 class AudioProcessorInterface; | |
| 27 } | |
| 28 | |
| 29 namespace content { | |
| 30 | |
| 31 class MediaStreamAudioProcessor; | |
| 32 class WebRtcAudioSinkAdapter; | |
| 33 class WebRtcLocalAudioTrack; | |
| 34 | |
| 35 // Provides an implementation of the webrtc::AudioTrackInterface that can be | |
| 36 // bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an | |
| 37 // adapter that sits between the media stream object graph and WebRtc's object | |
| 38 // graph and proxies between the two. | |
| 39 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter | |
| 40 : NON_EXPORTED_BASE( | |
| 41 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) { | |
| 42 public: | |
| 43 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create( | |
| 44 const std::string& label, | |
| 45 webrtc::AudioSourceInterface* track_source); | |
| 46 | |
| 47 WebRtcLocalAudioTrackAdapter( | |
| 48 const std::string& label, | |
| 49 webrtc::AudioSourceInterface* track_source, | |
| 50 scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner); | |
| 51 | |
| 52 ~WebRtcLocalAudioTrackAdapter() override; | |
| 53 | |
| 54 void Initialize(WebRtcLocalAudioTrack* owner); | |
| 55 | |
| 56 // Set the object that provides shared access to the current audio signal | |
| 57 // level. This method may only be called once, before the audio data flow | |
| 58 // starts, and before any calls to GetSignalLevel() might be made. | |
| 59 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level); | |
| 60 | |
| 61 // Method called by the WebRtcLocalAudioTrack to set the processor that | |
| 62 // applies signal processing on the data of the track. | |
| 63 // This class will keep a reference of the |processor|. | |
| 64 // Called on the main render thread. | |
| 65 // This method may only be called once, before the audio data flow starts, and | |
| 66 // before any calls to GetAudioProcessor() might be made. | |
| 67 void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor); | |
| 68 | |
| 69 // webrtc::MediaStreamTrack implementation. | |
| 70 std::string kind() const override; | |
| 71 bool set_enabled(bool enable) override; | |
| 72 | |
| 73 private: | |
| 74 // webrtc::AudioTrackInterface implementation. | |
| 75 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; | |
| 76 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; | |
| 77 bool GetSignalLevel(int* level) override; | |
| 78 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() | |
| 79 override; | |
| 80 webrtc::AudioSourceInterface* GetSource() const override; | |
| 81 | |
| 82 // Weak reference. | |
| 83 WebRtcLocalAudioTrack* owner_; | |
| 84 | |
| 85 // The source of the audio track which handles the audio constraints. | |
| 86 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. | |
| 87 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; | |
| 88 | |
| 89 // Libjingle's signaling thread. | |
| 90 const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_; | |
| 91 | |
| 92 // The audio processsor that applies audio processing on the data of audio | |
| 93 // track. This must be set before calls to GetAudioProcessor() are made. | |
| 94 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; | |
| 95 | |
| 96 // A vector of the peer connection sink adapters which receive the audio data | |
| 97 // from the audio track. | |
| 98 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_; | |
| 99 | |
| 100 // Thread-safe accessor to current audio signal level. This must be set | |
| 101 // before calls to GetSignalLevel() are made. | |
| 102 scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_; | |
| 103 }; | |
| 104 | |
| 105 } // namespace content | |
| 106 | |
| 107 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | |
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