e814a0dee0
This CL adds the following interfaces: * RtpTransportController * RtpTransport * RtpSender * RtpReceiver They're implemented on top of the "BaseChannel" object, which is normally used in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result of this, there are several limitations: * You can only have one of each type of sender and receiver (audio/video) on top of the same transport controller. * The sender/receiver with the same media type must use the same RTP transport. * You can't change the transport after creating the sender or receiver. * Some of the parameters aren't supported. Later, these "adapter" objects will be gradually replaced by real objects that don't have these limitations, as "BaseChannel", "MediaChannel" and related code is restructured. In this CL, we essentially have: ORTC adapter objects -> BaseChannel -> Media engine PeerConnection -> BaseChannel -> Media engine And later we hope to have simply: PeerConnection -> "Real" ORTC objects -> Media engine See the linked bug for more context. BUG=webrtc:7013 TBR=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2675173003 Cr-Commit-Position: refs/heads/master@{#16842}
78 lines
3.1 KiB
C++
78 lines
3.1 KiB
C++
/*
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* Copyright 2017 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://publications.ortc.org/2016/20161202/#rtcrtpsender*
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//
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// However, underneath the RtpSender is an RtpTransport, rather than a
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// DtlsTransport. This is to allow different types of RTP transports (besides
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// DTLS-SRTP) to be used.
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#ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
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#define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
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#include "webrtc/api/mediatypes.h"
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/ortc/rtptransportinterface.h"
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#include "webrtc/api/rtcerror.h"
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#include "webrtc/api/rtpparameters.h"
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namespace webrtc {
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// Note: Since sender capabilities may depend on how the OrtcFactory was
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// created, instead of a static "GetCapabilities" method on this interface,
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// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
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class OrtcRtpSenderInterface {
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public:
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virtual ~OrtcRtpSenderInterface() {}
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// Sets the source of media that will be sent by this sender.
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//
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// If Send has already been called, will immediately switch to sending this
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// track. If |track| is null, will stop sending media.
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//
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// Returns INVALID_PARAMETER error if an audio track is set on a video
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// RtpSender, or vice-versa.
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virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
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// Returns previously set (or constructed-with) track.
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
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// Once supported, will switch to sending media on a new transport. However,
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// this is not currently supported and will always return an error.
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virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
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// Returns previously set (or constructed-with) transport.
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virtual RtpTransportInterface* GetTransport() const = 0;
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// Start sending media with |parameters| (if |parameters| contains an active
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// encoding).
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//
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// There are no limitations to how the parameters can be changed after the
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// initial call to Send, as long as they're valid (for example, they can't
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// use the same payload type for two codecs).
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virtual RTCError Send(const RtpParameters& parameters) = 0;
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// Returns parameters that were last successfully passed into Send, or empty
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// parameters if that hasn't yet occurred.
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//
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// Note that for parameters that are described as having an "implementation
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// default" value chosen, GetParameters() will return those chosen defaults,
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// with the exception of SSRCs which have special behavior. See
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// rtpparameters.h for more details.
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virtual RtpParameters GetParameters() const = 0;
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// Audio or video sender?
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virtual cricket::MediaType GetKind() const = 0;
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// TODO(deadbeef): SSRC conflict signal.
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};
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} // namespace webrtc
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#endif // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
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