1/* 2 * libjingle 3 * Copyright 2015 Google Inc. 4 * 5 * Redistribution and use in source and binary forms, with or without 6 * modification, are permitted provided that the following conditions are met: 7 * 8 * 1. Redistributions of source code must retain the above copyright notice, 9 * this list of conditions and the following disclaimer. 10 * 2. Redistributions in binary form must reproduce the above copyright notice, 11 * this list of conditions and the following disclaimer in the documentation 12 * and/or other materials provided with the distribution. 13 * 3. The name of the author may not be used to endorse or promote products 14 * derived from this software without specific prior written permission. 15 * 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 26 */ 27 28// This file contains classes that implement RtpReceiverInterface. 29// An RtpReceiver associates a MediaStreamTrackInterface with an underlying 30// transport (provided by AudioProviderInterface/VideoProviderInterface) 31 32#ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_ 33#define TALK_APP_WEBRTC_RTPRECEIVER_H_ 34 35#include <string> 36 37#include "talk/app/webrtc/mediastreamprovider.h" 38#include "talk/app/webrtc/rtpreceiverinterface.h" 39#include "webrtc/base/basictypes.h" 40 41namespace webrtc { 42 43class AudioRtpReceiver : public ObserverInterface, 44 public AudioSourceInterface::AudioObserver, 45 public rtc::RefCountedObject<RtpReceiverInterface> { 46 public: 47 AudioRtpReceiver(AudioTrackInterface* track, 48 uint32_t ssrc, 49 AudioProviderInterface* provider); 50 51 virtual ~AudioRtpReceiver(); 52 53 // ObserverInterface implementation 54 void OnChanged() override; 55 56 // AudioSourceInterface::AudioObserver implementation 57 void OnSetVolume(double volume) override; 58 59 // RtpReceiverInterface implementation 60 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 61 return track_.get(); 62 } 63 64 std::string id() const override { return id_; } 65 66 void Stop() override; 67 68 private: 69 void Reconfigure(); 70 71 const std::string id_; 72 const rtc::scoped_refptr<AudioTrackInterface> track_; 73 const uint32_t ssrc_; 74 AudioProviderInterface* provider_; // Set to null in Stop(). 75 bool cached_track_enabled_; 76}; 77 78class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> { 79 public: 80 VideoRtpReceiver(VideoTrackInterface* track, 81 uint32_t ssrc, 82 VideoProviderInterface* provider); 83 84 virtual ~VideoRtpReceiver(); 85 86 // RtpReceiverInterface implementation 87 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { 88 return track_.get(); 89 } 90 91 std::string id() const override { return id_; } 92 93 void Stop() override; 94 95 private: 96 std::string id_; 97 rtc::scoped_refptr<VideoTrackInterface> track_; 98 uint32_t ssrc_; 99 VideoProviderInterface* provider_; 100}; 101 102} // namespace webrtc 103 104#endif // TALK_APP_WEBRTC_RTPRECEIVER_H_ 105