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| 1 // Copyright 2013 The Chromium Authors. All rights reserved. | 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 | 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_LOCAL_AUDIO_TRACK_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
| 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
| 7 | 7 |
| 8 #include <list> | 8 #include <list> |
| 9 #include <string> | 9 #include <string> |
| 10 | 10 |
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| 58 | 58 |
| 59 // Starts the local audio track. Called on the main render thread and | 59 // Starts the local audio track. Called on the main render thread and |
| 60 // should be called only once when audio track is created. | 60 // should be called only once when audio track is created. |
| 61 void Start(); | 61 void Start(); |
| 62 | 62 |
| 63 // Stops the local audio track. Called on the main render thread and | 63 // Stops the local audio track. Called on the main render thread and |
| 64 // should be called only once when audio track going away. | 64 // should be called only once when audio track going away. |
| 65 void Stop(); | 65 void Stop(); |
| 66 | 66 |
| 67 // Method called by the capturer to deliver the capture data. | 67 // Method called by the capturer to deliver the capture data. |
| 68 void Capture(media::AudioBus* audio_source, | 68 void Capture(const int16* audio_data, |
| 69 int audio_delay_milliseconds, | 69 int sample_rate, |
| 70 int volume, | 70 int number_of_channels, |
| 71 bool key_pressed); | 71 int number_of_frames); |
| 72 | 72 |
| 73 // Method called by the capturer to set the audio parameters used by source | 73 // Method called by the capturer to set the audio parameters used by source |
| 74 // of the capture data.. | 74 // of the capture data.. |
| 75 // Can be called on different user threads. | 75 // Can be called on different user threads. |
| 76 void SetCaptureFormat(const media::AudioParameters& params); | 76 void SetCaptureFormat(const media::AudioParameters& params); |
| 77 | 77 |
| 78 protected: | 78 protected: |
| 79 WebRtcLocalAudioTrack( | 79 WebRtcLocalAudioTrack( |
| 80 const std::string& label, | 80 const std::string& label, |
| 81 const scoped_refptr<WebRtcAudioCapturer>& capturer, | 81 const scoped_refptr<WebRtcAudioCapturer>& capturer, |
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| 118 base::ThreadChecker thread_checker_; | 118 base::ThreadChecker thread_checker_; |
| 119 | 119 |
| 120 // Protects |params_| and |sinks_|. | 120 // Protects |params_| and |sinks_|. |
| 121 mutable base::Lock lock_; | 121 mutable base::Lock lock_; |
| 122 | 122 |
| 123 // A vector of WebRtc VoE channels that the capturer sends datat to. | 123 // A vector of WebRtc VoE channels that the capturer sends datat to. |
| 124 std::vector<int> voe_channels_; | 124 std::vector<int> voe_channels_; |
| 125 | 125 |
| 126 bool need_audio_processing_; | 126 bool need_audio_processing_; |
| 127 | 127 |
| 128 // Buffers used for temporary storage during capture callbacks. | 128 media::AudioParameters params_; |
| 129 // Allocated during initialization. | |
| 130 class ConfiguredBuffer; | |
| 131 scoped_refptr<ConfiguredBuffer> buffer_; | |
| 132 | 129 |
| 133 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); | 130 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); |
| 134 }; | 131 }; |
| 135 | 132 |
| 136 } // namespace content | 133 } // namespace content |
| 137 | 134 |
| 138 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ | 135 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ |
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