| /* |
| * 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 interfaces for RtpSenders |
| // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface |
| |
| #ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ |
| #define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ |
| |
| #include <string> |
| |
| #include "talk/app/webrtc/proxy.h" |
| #include "talk/app/webrtc/mediastreaminterface.h" |
| #include "talk/session/media/mediasession.h" |
| #include "webrtc/base/refcount.h" |
| #include "webrtc/base/scoped_ref_ptr.h" |
| |
| namespace webrtc { |
| |
| class RtpSenderInterface : public rtc::RefCountInterface { |
| public: |
| // Returns true if successful in setting the track. |
| // Fails if an audio track is set on a video RtpSender, or vice-versa. |
| virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; |
| virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; |
| |
| // Used to set the SSRC of the sender, once a local description has been set. |
| // If |ssrc| is 0, this indiates that the sender should disconnect from the |
| // underlying transport (this occurs if the sender isn't seen in a local |
| // description). |
| virtual void SetSsrc(uint32_t ssrc) = 0; |
| virtual uint32_t ssrc() const = 0; |
| |
| // Audio or video sender? |
| virtual cricket::MediaType media_type() const = 0; |
| |
| // Not to be confused with "mid", this is a field we can temporarily use |
| // to uniquely identify a receiver until we implement Unified Plan SDP. |
| virtual std::string id() const = 0; |
| |
| // TODO(deadbeef): Support one sender having multiple stream ids. |
| virtual void set_stream_id(const std::string& stream_id) = 0; |
| virtual std::string stream_id() const = 0; |
| |
| virtual void Stop() = 0; |
| |
| protected: |
| virtual ~RtpSenderInterface() {} |
| }; |
| |
| // Define proxy for RtpSenderInterface. |
| BEGIN_PROXY_MAP(RtpSender) |
| PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) |
| PROXY_METHOD1(void, SetSsrc, uint32_t) |
| PROXY_CONSTMETHOD0(uint32_t, ssrc) |
| PROXY_CONSTMETHOD0(cricket::MediaType, media_type) |
| PROXY_CONSTMETHOD0(std::string, id) |
| PROXY_METHOD1(void, set_stream_id, const std::string&) |
| PROXY_CONSTMETHOD0(std::string, stream_id) |
| PROXY_METHOD0(void, Stop) |
| END_PROXY() |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ |