Move talk/app/webrtc to webrtc/api

talk/app/webrtc/rtpsenderinterface.h

-/*
- * 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/mediastreaminterface.h"
-#include "talk/app/webrtc/proxy.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_