106 lines
4.2 KiB
C
106 lines
4.2 KiB
C
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/*
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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_RTP_SENDER_INTERFACE_H_
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#define API_RTP_SENDER_INTERFACE_H_
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#include <string>
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#include <vector>
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#include "api/crypto/frame_encryptor_interface.h"
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#include "api/dtls_transport_interface.h"
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#include "api/dtmf_sender_interface.h"
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#include "api/frame_transformer_interface.h"
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#include "api/media_stream_interface.h"
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#include "api/media_types.h"
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#include "api/rtc_error.h"
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#include "api/rtp_parameters.h"
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#include "api/scoped_refptr.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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class RTC_EXPORT 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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// The dtlsTransport attribute exposes the DTLS transport on which the
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// media is sent. It may be null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
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// TODO(https://bugs.webrtc.org/907849) remove default implementation
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virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
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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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// Sets the IDs of the media streams 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 void SetStreams(const std::vector<std::string>& stream_ids) {}
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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() const = 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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virtual void SetEncoderToPacketizerFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
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protected:
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~RtpSenderInterface() override = default;
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};
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} // namespace webrtc
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#endif // API_RTP_SENDER_INTERFACE_H_
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