| /* |
| * libjingle |
| * Copyright 2015 Google Inc. |
| * |
| * Redistribution and use in source and binary forms, with or without |
| * modification, are permitted provided that the following conditions are met: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| // This file contains classes that implement RtpReceiverInterface. |
| // An RtpReceiver associates a MediaStreamTrackInterface with an underlying |
| // transport (provided by AudioProviderInterface/VideoProviderInterface) |
| |
| #ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_ |
| #define TALK_APP_WEBRTC_RTPRECEIVER_H_ |
| |
| #include <string> |
| |
| #include "talk/app/webrtc/mediastreamprovider.h" |
| #include "talk/app/webrtc/rtpreceiverinterface.h" |
| #include "webrtc/base/basictypes.h" |
| |
| namespace webrtc { |
| |
| class AudioRtpReceiver : public ObserverInterface, |
| public AudioSourceInterface::AudioObserver, |
| public rtc::RefCountedObject<RtpReceiverInterface> { |
| public: |
| AudioRtpReceiver(AudioTrackInterface* track, |
| uint32_t ssrc, |
| AudioProviderInterface* provider); |
| |
| virtual ~AudioRtpReceiver(); |
| |
| // ObserverInterface implementation |
| void OnChanged() override; |
| |
| // AudioSourceInterface::AudioObserver implementation |
| void OnSetVolume(double volume) override; |
| |
| // 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(); |
| |
| std::string id_; |
| rtc::scoped_refptr<AudioTrackInterface> track_; |
| uint32_t ssrc_; |
| AudioProviderInterface* provider_; |
| bool cached_track_enabled_; |
| }; |
| |
| class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> { |
| public: |
| VideoRtpReceiver(VideoTrackInterface* track, |
| uint32_t ssrc, |
| VideoProviderInterface* provider); |
| |
| virtual ~VideoRtpReceiver(); |
| |
| // 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_; |
| rtc::scoped_refptr<VideoTrackInterface> track_; |
| uint32_t ssrc_; |
| VideoProviderInterface* provider_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_RTPRECEIVER_H_ |