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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
77 lines
3.1 KiB
C++
77 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/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 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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