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}
85 lines
3.5 KiB
C++
85 lines
3.5 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 RtpReceivers:
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// http://publications.ortc.org/2016/20161202/#rtcrtpreceiver*
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//
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// However, underneath the RtpReceiver 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_ORTCRTPRECEIVERINTERFACE_H_
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#define WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/mediatypes.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 receiver 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 "GetRtpReceiverCapabilities" method on the OrtcFactory.
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class OrtcRtpReceiverInterface {
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public:
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virtual ~OrtcRtpReceiverInterface() {}
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// Returns a track representing the media received by this receiver.
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//
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// Currently, this will return null until Receive has been successfully
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// called. Also, a new track will be created every time the primary SSRC
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// changes.
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//
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// If encodings are removed, GetTrack will return null. Though deactivating
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// an encoding (setting |active| to false) will not do this.
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//
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// In the future, these limitations will be fixed, and GetTrack will return
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// the same track for the lifetime of the RtpReceiver. So it's not
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// recommended to write code that depends on this non-standard behavior.
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
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// Once supported, will switch to receiving media on a new transport.
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// However, 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 receiving media with |parameters| (if |parameters| contains an
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// active 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 Receive, 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 Receive(const RtpParameters& parameters) = 0;
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// Returns parameters that were last successfully passed into Receive, or
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// empty 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 receiver?
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//
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// Once GetTrack() starts always returning a track, this method will be
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// redundant, as one can call "GetTrack()->kind()". However, it's still a
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// nice convenience, and is symmetric with OrtcRtpSenderInterface::GetKind.
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virtual cricket::MediaType GetKind() const = 0;
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// TODO(deadbeef): GetContributingSources
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};
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} // namespace webrtc
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#endif // WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
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