blob: a0fc208a58f8503a1dbf19048b8abae65feebc04 [file] [log] [blame]
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
// This file contains interfaces for RtpSenders:
// However, underneath the RtpSender is an RtpTransport, rather than a
// DtlsTransport. This is to allow different types of RTP transports (besides
// DTLS-SRTP) to be used.
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/api/rtpparameters.h"
namespace webrtc {
// Note: Since sender capabilities may depend on how the OrtcFactory was
// created, instead of a static "GetCapabilities" method on this interface,
// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
class OrtcRtpSenderInterface {
virtual ~OrtcRtpSenderInterface() {}
// Sets the source of media that will be sent by this sender.
// If Send has already been called, will immediately switch to sending this
// track. If |track| is null, will stop sending media.
// Returns INVALID_PARAMETER error if an audio track is set on a video
// RtpSender, or vice-versa.
virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
// Returns previously set (or constructed-with) track.
virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
// Once supported, will switch to sending media on a new transport. However,
// this is not currently supported and will always return an error.
virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
// Returns previously set (or constructed-with) transport.
virtual RtpTransportInterface* GetTransport() const = 0;
// Start sending media with |parameters| (if |parameters| contains an active
// encoding).
// There are no limitations to how the parameters can be changed after the
// initial call to Send, as long as they're valid (for example, they can't
// use the same payload type for two codecs).
virtual RTCError Send(const RtpParameters& parameters) = 0;
// Returns parameters that were last successfully passed into Send, or empty
// parameters if that hasn't yet occurred.
// Note that for parameters that are described as having an "implementation
// default" value chosen, GetParameters() will return those chosen defaults,
// with the exception of SSRCs which have special behavior. See
// rtpparameters.h for more details.
virtual RtpParameters GetParameters() const = 0;
// Audio or video sender?
virtual cricket::MediaType GetKind() const = 0;
// TODO(deadbeef): SSRC conflict signal.
} // namespace webrtc