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892acf01f6
This will later allow simulcast to be set up without any SDP manipulation. Currently limited to only one layer as the SDP generated is not spec compliant and more work is required to support simulcast. Initial encoding parameters are deferred and applied when the ssrc is set on the sender. This allows parameters to be changed before negotiation is completed. Bug: webrtc:7600 Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d Reviewed-on: https://webrtc-review.googlesource.com/95488 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24917}
112 lines
4.5 KiB
C++
112 lines
4.5 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 API_RTPSENDERINTERFACE_H_
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#define API_RTPSENDERINTERFACE_H_
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#include <string>
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#include <vector>
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#include "api/crypto/frameencryptorinterface.h"
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#include "api/dtmfsenderinterface.h"
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#include "api/mediastreaminterface.h"
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#include "api/mediatypes.h"
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#include "api/proxy.h"
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#include "api/rtcerror.h"
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#include "api/rtpparameters.h"
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#include "rtc_base/deprecation.h"
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#include "rtc_base/refcount.h"
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#include "rtc_base/scoped_ref_ptr.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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// Returns primary SSRC used by this sender for sending media.
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// Returns 0 if not yet determined.
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// TODO(deadbeef): Change to absl::optional.
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// TODO(deadbeef): Remove? With GetParameters this should be redundant.
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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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// Returns a list of media stream ids associated with this sender's track.
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// These are signalled in the SDP so that the remote side can associate
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// tracks.
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virtual std::vector<std::string> stream_ids() const = 0;
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// Returns the list of encoding parameters that will be applied when the SDP
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// local description is set. These initial encoding parameters can be set by
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// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
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// TODO(orphis): Make it pure virtual once Chrome has updated
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virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
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virtual RtpParameters GetParameters() = 0;
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// Note that only a subset of the parameters can currently be changed. See
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// rtpparameters.h
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// The encodings are in increasing quality order for simulcast.
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virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
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// Returns null for a video sender.
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virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
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// Sets a user defined frame encryptor that will encrypt the entire frame
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// before it is sent across the network. This will encrypt the entire frame
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// using the user provided encryption mechanism regardless of whether SRTP is
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// enabled or not.
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virtual void SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
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// Returns a pointer to the frame encryptor set previously by the
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// user. This can be used to update the state of the object.
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virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
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protected:
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~RtpSenderInterface() override = default;
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};
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// Define proxy for RtpSenderInterface.
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// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
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// are called on is an implementation detail.
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BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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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_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_CONSTMETHOD0(std::vector<std::string>, stream_ids)
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PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
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PROXY_METHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
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PROXY_METHOD1(void,
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SetFrameEncryptor,
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rtc::scoped_refptr<FrameEncryptorInterface>);
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
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GetFrameEncryptor);
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END_PROXY_MAP()
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} // namespace webrtc
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#endif // API_RTPSENDERINTERFACE_H_
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