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| 1 // Copyright 2014 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
| 7 | 7 |
| 8 #include <vector> | 8 #include <vector> |
| 9 | 9 |
| 10 #include "base/memory/ref_counted.h" | 10 #include "base/memory/ref_counted.h" |
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| 64 bool set_enabled(bool enable) override; | 64 bool set_enabled(bool enable) override; |
| 65 | 65 |
| 66 private: | 66 private: |
| 67 // webrtc::AudioTrackInterface implementation. | 67 // webrtc::AudioTrackInterface implementation. |
| 68 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; | 68 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; |
| 69 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; | 69 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; |
| 70 bool GetSignalLevel(int* level) override; | 70 bool GetSignalLevel(int* level) override; |
| 71 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() | 71 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() |
| 72 override; | 72 override; |
| 73 webrtc::AudioSourceInterface* GetSource() const override; | 73 webrtc::AudioSourceInterface* GetSource() const override; |
| 74 cricket::AudioRenderer* GetRenderer() override; | |
| 75 | 74 |
| 76 // Weak reference. | 75 // Weak reference. |
| 77 WebRtcLocalAudioTrack* owner_; | 76 WebRtcLocalAudioTrack* owner_; |
| 78 | 77 |
| 79 // The source of the audio track which handles the audio constraints. | 78 // The source of the audio track which handles the audio constraints. |
| 80 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. | 79 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. |
| 81 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; | 80 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; |
| 82 | 81 |
| 83 // Libjingle's signaling thread. | 82 // Libjingle's signaling thread. |
| 84 const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_; | 83 const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_; |
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| 101 base::ThreadChecker signaling_thread_checker_; | 100 base::ThreadChecker signaling_thread_checker_; |
| 102 base::ThreadChecker capture_thread_; | 101 base::ThreadChecker capture_thread_; |
| 103 | 102 |
| 104 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|. | 103 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|. |
| 105 mutable base::Lock lock_; | 104 mutable base::Lock lock_; |
| 106 }; | 105 }; |
| 107 | 106 |
| 108 } // namespace content | 107 } // namespace content |
| 109 | 108 |
| 110 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ | 109 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ |
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