dc1c62cd30
This change allows the application to limit the bitrate of the outgoing audio and video streams at runtime. The API roughly follows the WebRTC API draft, defining the RTCRtpParameters structure witn exactly one encoding (simulcast streams are not exposed in the API for now). (https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters) BUG= Review URL: https://codereview.webrtc.org/1788583004 Cr-Commit-Position: refs/heads/master@{#12025}
80 lines
2.8 KiB
C++
80 lines
2.8 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/api/rtpparameters.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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virtual RtpParameters GetParameters() const = 0;
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virtual bool SetParameters(const RtpParameters& parameters) = 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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PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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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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