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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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