1 /* 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 // This file contains interfaces for RtpSenders 12 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface 13 14 #ifndef API_RTP_SENDER_INTERFACE_H_ 15 #define API_RTP_SENDER_INTERFACE_H_ 16 17 #include <memory> 18 #include <string> 19 #include <vector> 20 21 #include "absl/functional/any_invocable.h" 22 #include "api/crypto/frame_encryptor_interface.h" 23 #include "api/dtls_transport_interface.h" 24 #include "api/dtmf_sender_interface.h" 25 #include "api/frame_transformer_interface.h" 26 #include "api/media_stream_interface.h" 27 #include "api/media_types.h" 28 #include "api/rtc_error.h" 29 #include "api/rtp_parameters.h" 30 #include "api/scoped_refptr.h" 31 #include "api/video_codecs/video_encoder_factory.h" 32 #include "rtc_base/ref_count.h" 33 #include "rtc_base/system/rtc_export.h" 34 35 namespace webrtc { 36 37 using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>; 38 39 class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface { 40 public: 41 // Returns true if successful in setting the track. 42 // Fails if an audio track is set on a video RtpSender, or vice-versa. 43 virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; 44 virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; 45 46 // The dtlsTransport attribute exposes the DTLS transport on which the 47 // media is sent. It may be null. 48 // https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport 49 virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0; 50 51 // Returns primary SSRC used by this sender for sending media. 52 // Returns 0 if not yet determined. 53 // TODO(deadbeef): Change to absl::optional. 54 // TODO(deadbeef): Remove? With GetParameters this should be redundant. 55 virtual uint32_t ssrc() const = 0; 56 57 // Audio or video sender? 58 virtual cricket::MediaType media_type() const = 0; 59 60 // Not to be confused with "mid", this is a field we can temporarily use 61 // to uniquely identify a receiver until we implement Unified Plan SDP. 62 virtual std::string id() const = 0; 63 64 // Returns a list of media stream ids associated with this sender's track. 65 // These are signalled in the SDP so that the remote side can associate 66 // tracks. 67 virtual std::vector<std::string> stream_ids() const = 0; 68 69 // Sets the IDs of the media streams associated with this sender's track. 70 // These are signalled in the SDP so that the remote side can associate 71 // tracks. 72 virtual void SetStreams(const std::vector<std::string>& stream_ids) = 0; 73 74 // Returns the list of encoding parameters that will be applied when the SDP 75 // local description is set. These initial encoding parameters can be set by 76 // PeerConnection::AddTransceiver, and later updated with Get/SetParameters. 77 // TODO(orphis): Make it pure virtual once Chrome has updated 78 virtual std::vector<RtpEncodingParameters> init_send_encodings() const = 0; 79 80 virtual RtpParameters GetParameters() const = 0; 81 // Note that only a subset of the parameters can currently be changed. See 82 // rtpparameters.h 83 // The encodings are in increasing quality order for simulcast. 84 virtual RTCError SetParameters(const RtpParameters& parameters) = 0; 85 virtual void SetParametersAsync(const RtpParameters& parameters, 86 SetParametersCallback callback); 87 88 // Returns null for a video sender. 89 virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0; 90 91 // Sets a user defined frame encryptor that will encrypt the entire frame 92 // before it is sent across the network. This will encrypt the entire frame 93 // using the user provided encryption mechanism regardless of whether SRTP is 94 // enabled or not. 95 virtual void SetFrameEncryptor( 96 rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0; 97 98 // Returns a pointer to the frame encryptor set previously by the 99 // user. This can be used to update the state of the object. 100 virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() 101 const = 0; 102 103 virtual void SetEncoderToPacketizerFrameTransformer( 104 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0; 105 106 // Sets a user defined encoder selector. 107 // Overrides selector that is (optionally) provided by VideoEncoderFactory. 108 virtual void SetEncoderSelector( 109 std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface> 110 encoder_selector) = 0; 111 112 // TODO(crbug.com/1354101): make pure virtual again after Chrome roll. GenerateKeyFrame(const std::vector<std::string> & rids)113 virtual RTCError GenerateKeyFrame(const std::vector<std::string>& rids) { 114 return RTCError::OK(); 115 } 116 117 protected: 118 ~RtpSenderInterface() override = default; 119 }; 120 121 } // namespace webrtc 122 123 #endif // API_RTP_SENDER_INTERFACE_H_ 124