 Chromium Code Reviews
 Chromium Code Reviews Issue 1397123003:
  Add AudioSendStream to Call.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master
    
  
    Issue 1397123003:
  Add AudioSendStream to Call.  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master| Index: webrtc/audio/audio_send_stream.h | 
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..c8fd70ff9bc7602ad284d4d944e46ea8ac196d4a | 
| --- /dev/null | 
| +++ b/webrtc/audio/audio_send_stream.h | 
| @@ -0,0 +1,44 @@ | 
| +/* | 
| + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 
| + * | 
| + * Use of this source code is governed by a BSD-style license | 
| + * that can be found in the LICENSE file in the root of the source | 
| + * tree. An additional intellectual property rights grant can be found | 
| + * in the file PATENTS. All contributing project authors may | 
| + * be found in the AUTHORS file in the root of the source tree. | 
| + */ | 
| + | 
| +#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 
| +#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ | 
| + | 
| +#include "webrtc/audio_send_stream.h" | 
| + | 
| +namespace webrtc { | 
| + | 
| 
pbos-webrtc
2015/10/15 15:32:11
remove empty line
 
the sun
2015/10/16 08:47:05
Done. Thanks. Important.
 | 
| +namespace internal { | 
| + | 
| +class AudioSendStream : public webrtc::AudioSendStream { | 
| + public: | 
| + explicit AudioSendStream(const webrtc::AudioSendStream::Config& config); | 
| + ~AudioSendStream() override; | 
| + | 
| + // webrtc::SendStream implementation. | 
| + void Start() override; | 
| + void Stop() override; | 
| + void SignalNetworkState(NetworkState state) override; | 
| + bool DeliverRtcp(const uint8_t* packet, size_t length) override; | 
| + | 
| + // webrtc::AudioSendStream implementation. | 
| + webrtc::AudioSendStream::Stats GetStats() const override; | 
| + | 
| + const webrtc::AudioSendStream::Config& config() const { | 
| + return config_; | 
| + } | 
| + | 
| + private: | 
| + const webrtc::AudioSendStream::Config config_; | 
| +}; | 
| +} // namespace internal | 
| +} // namespace webrtc | 
| + | 
| +#endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_ |