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/*
* Copyright 2015 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
// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
#ifndef API_RTP_SENDER_INTERFACE_H_
#define API_RTP_SENDER_INTERFACE_H_
#include <string>
#include <vector>
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/dtmf_sender_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/proxy.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
public:
// Returns true if successful in setting the track.
// Fails if an audio track is set on a video RtpSender, or vice-versa.
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
// The dtlsTransport attribute exposes the DTLS transport on which the
// media is sent. It may be null.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
// TODO(https://bugs.webrtc.org/907849) remove default implementation
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// Returns primary SSRC used by this sender for sending media.
// Returns 0 if not yet determined.
// TODO(deadbeef): Change to absl::optional.
// TODO(deadbeef): Remove? With GetParameters this should be redundant.
virtual uint32_t ssrc() const = 0;
// Audio or video sender?
virtual cricket::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
// Returns a list of media stream ids associated with this sender's track.
// These are signalled in the SDP so that the remote side can associate
// tracks.
virtual std::vector<std::string> stream_ids() const = 0;
// Sets the IDs of the media streams associated with this sender's track.
// These are signalled in the SDP so that the remote side can associate
// tracks.
virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
// Returns the list of encoding parameters that will be applied when the SDP
// local description is set. These initial encoding parameters can be set by
// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
// TODO(orphis): Make it pure virtual once Chrome has updated
virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
virtual RtpParameters GetParameters() const = 0;
// Note that only a subset of the parameters can currently be changed. See
// rtpparameters.h
// The encodings are in increasing quality order for simulcast.
virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
// Returns null for a video sender.
virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
// Sets a user defined frame encryptor that will encrypt the entire frame
// before it is sent across the network. This will encrypt the entire frame
// using the user provided encryption mechanism regardless of whether SRTP is
// enabled or not.
virtual void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
// Returns a pointer to the frame encryptor set previously by the
// user. This can be used to update the state of the object.
virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
virtual void SetEncoderToPacketizerFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
protected:
~RtpSenderInterface() override = default;
};
// Define proxy for RtpSenderInterface.
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
// are called on is an implementation detail.
BEGIN_SIGNALING_PROXY_MAP(RtpSender)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(uint32_t, ssrc)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender)
PROXY_METHOD1(void,
SetFrameEncryptor,
rtc::scoped_refptr<FrameEncryptorInterface>)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
GetFrameEncryptor)
PROXY_METHOD1(void, SetStreams, const std::vector<std::string>&)
PROXY_METHOD1(void,
SetEncoderToPacketizerFrameTransformer,
rtc::scoped_refptr<FrameTransformerInterface>)
END_PROXY_MAP()
} // namespace webrtc
#endif // API_RTP_SENDER_INTERFACE_H_
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