| /* | 
 |  *  Copyright 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. | 
 |  */ | 
 |  | 
 | // This file contains classes that implement RtpReceiverInterface. | 
 | // An RtpReceiver associates a MediaStreamTrackInterface with an underlying | 
 | // transport (provided by AudioProviderInterface/VideoProviderInterface) | 
 |  | 
 | #ifndef WEBRTC_API_RTPRECEIVER_H_ | 
 | #define WEBRTC_API_RTPRECEIVER_H_ | 
 |  | 
 | #include <string> | 
 |  | 
 | #include "webrtc/api/mediastreamprovider.h" | 
 | #include "webrtc/api/rtpreceiverinterface.h" | 
 | #include "webrtc/api/remoteaudiosource.h" | 
 | #include "webrtc/api/videotracksource.h" | 
 | #include "webrtc/base/basictypes.h" | 
 | #include "webrtc/media/base/videobroadcaster.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | class AudioRtpReceiver : public ObserverInterface, | 
 |                          public AudioSourceInterface::AudioObserver, | 
 |                          public rtc::RefCountedObject<RtpReceiverInterface> { | 
 |  public: | 
 |   AudioRtpReceiver(MediaStreamInterface* stream, | 
 |                    const std::string& track_id, | 
 |                    uint32_t ssrc, | 
 |                    AudioProviderInterface* provider); | 
 |  | 
 |   virtual ~AudioRtpReceiver(); | 
 |  | 
 |   // ObserverInterface implementation | 
 |   void OnChanged() override; | 
 |  | 
 |   // AudioSourceInterface::AudioObserver implementation | 
 |   void OnSetVolume(double volume) override; | 
 |  | 
 |   rtc::scoped_refptr<AudioTrackInterface> audio_track() const { | 
 |     return track_.get(); | 
 |   } | 
 |  | 
 |   // RtpReceiverInterface implementation | 
 |   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 
 |     return track_.get(); | 
 |   } | 
 |  | 
 |   std::string id() const override { return id_; } | 
 |  | 
 |   void Stop() override; | 
 |  | 
 |  private: | 
 |   void Reconfigure(); | 
 |  | 
 |   const std::string id_; | 
 |   const uint32_t ssrc_; | 
 |   AudioProviderInterface* provider_;  // Set to null in Stop(). | 
 |   const rtc::scoped_refptr<AudioTrackInterface> track_; | 
 |   bool cached_track_enabled_; | 
 | }; | 
 |  | 
 | class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> { | 
 |  public: | 
 |   VideoRtpReceiver(MediaStreamInterface* stream, | 
 |                    const std::string& track_id, | 
 |                    rtc::Thread* worker_thread, | 
 |                    uint32_t ssrc, | 
 |                    VideoProviderInterface* provider); | 
 |  | 
 |   virtual ~VideoRtpReceiver(); | 
 |  | 
 |   rtc::scoped_refptr<VideoTrackInterface> video_track() const { | 
 |     return track_.get(); | 
 |   } | 
 |  | 
 |   // RtpReceiverInterface implementation | 
 |   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | 
 |     return track_.get(); | 
 |   } | 
 |  | 
 |   std::string id() const override { return id_; } | 
 |  | 
 |   void Stop() override; | 
 |  | 
 |  private: | 
 |   std::string id_; | 
 |   uint32_t ssrc_; | 
 |   VideoProviderInterface* provider_; | 
 |   // |broadcaster_| is needed since the decoder can only handle one sink. | 
 |   // It might be better if the decoder can handle multiple sinks and consider | 
 |   // the VideoSinkWants. | 
 |   rtc::VideoBroadcaster broadcaster_; | 
 |   // |source_| is held here to be able to change the state of the source when | 
 |   // the VideoRtpReceiver is stopped. | 
 |   rtc::scoped_refptr<VideoTrackSource> source_; | 
 |   rtc::scoped_refptr<VideoTrackInterface> track_; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // WEBRTC_API_RTPRECEIVER_H_ |