9b8df25c73
The libjingle_p2p target is renamed to rtc_pc. The libjingle_p2p_unittest test will be renamed in a separate follow-up CL, to make it possible to run all trybots successfully for this CL. BUG=webrtc:5419 R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1691463002 . Cr-Commit-Position: refs/heads/master@{#11592}
74 lines
2.5 KiB
C++
74 lines
2.5 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains interfaces for RtpSenders
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// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
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#ifndef WEBRTC_API_RTPSENDERINTERFACE_H_
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#define WEBRTC_API_RTPSENDERINTERFACE_H_
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#include <string>
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/proxy.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/pc/mediasession.h"
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namespace webrtc {
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class RtpSenderInterface : public rtc::RefCountInterface {
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public:
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// Returns true if successful in setting the track.
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// Fails if an audio track is set on a video RtpSender, or vice-versa.
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virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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// Used to set the SSRC of the sender, once a local description has been set.
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// If |ssrc| is 0, this indiates that the sender should disconnect from the
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// underlying transport (this occurs if the sender isn't seen in a local
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// description).
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virtual void SetSsrc(uint32_t ssrc) = 0;
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virtual uint32_t ssrc() const = 0;
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// Audio or video sender?
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virtual cricket::MediaType media_type() const = 0;
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// Not to be confused with "mid", this is a field we can temporarily use
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// to uniquely identify a receiver until we implement Unified Plan SDP.
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virtual std::string id() const = 0;
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// TODO(deadbeef): Support one sender having multiple stream ids.
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virtual void set_stream_id(const std::string& stream_id) = 0;
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virtual std::string stream_id() const = 0;
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virtual void Stop() = 0;
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protected:
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virtual ~RtpSenderInterface() {}
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};
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// Define proxy for RtpSenderInterface.
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BEGIN_PROXY_MAP(RtpSender)
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PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_METHOD1(void, SetSsrc, uint32_t)
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PROXY_CONSTMETHOD0(uint32_t, ssrc)
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PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
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PROXY_CONSTMETHOD0(std::string, id)
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PROXY_METHOD1(void, set_stream_id, const std::string&)
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PROXY_CONSTMETHOD0(std::string, stream_id)
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PROXY_METHOD0(void, Stop)
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END_PROXY()
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} // namespace webrtc
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#endif // WEBRTC_API_RTPSENDERINTERFACE_H_
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