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| 1 // Copyright 2016 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 REMOTING_PROTOCOL_WEBRTC_AUDIO_SINK_ADAPTER_H_ |
| 6 #define REMOTING_PROTOCOL_WEBRTC_AUDIO_SINK_ADAPTER_H_ |
| 7 |
| 8 #include "base/memory/ref_counted.h" |
| 9 #include "base/memory/weak_ptr.h" |
| 10 #include "third_party/webrtc/api/mediastreaminterface.h" |
| 11 |
| 12 namespace remoting { |
| 13 namespace protocol { |
| 14 |
| 15 class AudioStub; |
| 16 |
| 17 class WebrtcAudioSinkAdapter : public webrtc::AudioTrackSinkInterface { |
| 18 public: |
| 19 WebrtcAudioSinkAdapter(scoped_refptr<webrtc::MediaStreamInterface> stream, |
| 20 base::WeakPtr<AudioStub> audio_stub); |
| 21 ~WebrtcAudioSinkAdapter() override; |
| 22 |
| 23 void OnData(const void* audio_data, |
| 24 int bits_per_sample, |
| 25 int sample_rate, |
| 26 size_t number_of_channels, |
| 27 size_t number_of_frames) override; |
| 28 |
| 29 private: |
| 30 scoped_refptr<webrtc::MediaStreamInterface> media_stream_; |
| 31 scoped_refptr<webrtc::AudioTrackInterface> audio_track_; |
| 32 |
| 33 base::WeakPtr<AudioStub> audio_stub_; |
| 34 }; |
| 35 |
| 36 } // namespace protocol |
| 37 } // namespace remoting |
| 38 |
| 39 #endif // REMOTING_PROTOCOL_WEBRTC_AUDIO_SINK_ADAPTER_H_ |
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