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 interfaces for RtpSenders 29 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface 30 31 #ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ 32 #define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ 33 34 #include <string> 35 36 #include "talk/app/webrtc/proxy.h" 37 #include "talk/app/webrtc/mediastreaminterface.h" 38 #include "talk/session/media/mediasession.h" 39 #include "webrtc/base/refcount.h" 40 #include "webrtc/base/scoped_ref_ptr.h" 41 42 namespace webrtc { 43 44 class RtpSenderInterface : public rtc::RefCountInterface { 45 public: 46 // Returns true if successful in setting the track. 47 // Fails if an audio track is set on a video RtpSender, or vice-versa. 48 virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; 49 virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; 50 51 // Used to set the SSRC of the sender, once a local description has been set. 52 // If |ssrc| is 0, this indiates that the sender should disconnect from the 53 // underlying transport (this occurs if the sender isn't seen in a local 54 // description). 55 virtual void SetSsrc(uint32_t ssrc) = 0; 56 virtual uint32_t ssrc() const = 0; 57 58 // Audio or video sender? 59 virtual cricket::MediaType media_type() const = 0; 60 61 // Not to be confused with "mid", this is a field we can temporarily use 62 // to uniquely identify a receiver until we implement Unified Plan SDP. 63 virtual std::string id() const = 0; 64 65 // TODO(deadbeef): Support one sender having multiple stream ids. 66 virtual void set_stream_id(const std::string& stream_id) = 0; 67 virtual std::string stream_id() const = 0; 68 69 virtual void Stop() = 0; 70 71 protected: 72 virtual ~RtpSenderInterface() {} 73 }; 74 75 // Define proxy for RtpSenderInterface. 76 BEGIN_PROXY_MAP(RtpSender) 77 PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) 78 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) 79 PROXY_METHOD1(void, SetSsrc, uint32_t) 80 PROXY_CONSTMETHOD0(uint32_t, ssrc) 81 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) 82 PROXY_CONSTMETHOD0(std::string, id) 83 PROXY_METHOD1(void, set_stream_id, const std::string&) 84 PROXY_CONSTMETHOD0(std::string, stream_id) 85 PROXY_METHOD0(void, Stop) 86 END_PROXY() 87 88 } // namespace webrtc 89 90 #endif // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ 91
