blob: 7c94c216e64efbeb5c4628bebe6925a6f5023e15 [file] [log] [blame]
* 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
#include <string>
#include <vector>
#include "api/crypto/frameencryptorinterface.h"
#include "api/dtmfsenderinterface.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/proxy.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
class RtpSenderInterface : public rtc::RefCountInterface {
// 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;
// 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;
// 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() = 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;
~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.
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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_METHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
} // namespace webrtc