1 /* 2 * Copyright 2017 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 #ifndef API_RTP_TRANSCEIVER_INTERFACE_H_ 12 #define API_RTP_TRANSCEIVER_INTERFACE_H_ 13 14 #include <string> 15 #include <vector> 16 17 #include "absl/types/optional.h" 18 #include "api/array_view.h" 19 #include "api/media_types.h" 20 #include "api/rtp_parameters.h" 21 #include "api/rtp_receiver_interface.h" 22 #include "api/rtp_sender_interface.h" 23 #include "api/rtp_transceiver_direction.h" 24 #include "api/scoped_refptr.h" 25 #include "rtc_base/ref_count.h" 26 #include "rtc_base/system/rtc_export.h" 27 28 namespace webrtc { 29 30 // Structure for initializing an RtpTransceiver in a call to 31 // PeerConnectionInterface::AddTransceiver. 32 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit 33 struct RTC_EXPORT RtpTransceiverInit final { 34 RtpTransceiverInit(); 35 RtpTransceiverInit(const RtpTransceiverInit&); 36 ~RtpTransceiverInit(); 37 // Direction of the RtpTransceiver. See RtpTransceiverInterface::direction(). 38 RtpTransceiverDirection direction = RtpTransceiverDirection::kSendRecv; 39 40 // The added RtpTransceiver will be added to these streams. 41 std::vector<std::string> stream_ids; 42 43 // TODO(bugs.webrtc.org/7600): Not implemented. 44 std::vector<RtpEncodingParameters> send_encodings; 45 }; 46 47 // The RtpTransceiverInterface maps to the RTCRtpTransceiver defined by the 48 // WebRTC specification. A transceiver represents a combination of an RtpSender 49 // and an RtpReceiver than share a common mid. As defined in JSEP, an 50 // RtpTransceiver is said to be associated with a media description if its mid 51 // property is non-null; otherwise, it is said to be disassociated. 52 // JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24 53 // 54 // Note that RtpTransceivers are only supported when using PeerConnection with 55 // Unified Plan SDP. 56 // 57 // This class is thread-safe. 58 // 59 // WebRTC specification for RTCRtpTransceiver, the JavaScript analog: 60 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver 61 class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface { 62 public: 63 // Media type of the transceiver. Any sender(s)/receiver(s) will have this 64 // type as well. 65 virtual cricket::MediaType media_type() const = 0; 66 67 // The mid attribute is the mid negotiated and present in the local and 68 // remote descriptions. Before negotiation is complete, the mid value may be 69 // null. After rollbacks, the value may change from a non-null value to null. 70 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid 71 virtual absl::optional<std::string> mid() const = 0; 72 73 // The sender attribute exposes the RtpSender corresponding to the RTP media 74 // that may be sent with the transceiver's mid. The sender is always present, 75 // regardless of the direction of media. 76 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender 77 virtual rtc::scoped_refptr<RtpSenderInterface> sender() const = 0; 78 79 // The receiver attribute exposes the RtpReceiver corresponding to the RTP 80 // media that may be received with the transceiver's mid. The receiver is 81 // always present, regardless of the direction of media. 82 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver 83 virtual rtc::scoped_refptr<RtpReceiverInterface> receiver() const = 0; 84 85 // The stopped attribute indicates that the sender of this transceiver will no 86 // longer send, and that the receiver will no longer receive. It is true if 87 // either stop has been called or if setting the local or remote description 88 // has caused the RtpTransceiver to be stopped. 89 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped 90 virtual bool stopped() const = 0; 91 92 // The direction attribute indicates the preferred direction of this 93 // transceiver, which will be used in calls to CreateOffer and CreateAnswer. 94 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction 95 virtual RtpTransceiverDirection direction() const = 0; 96 97 // Sets the preferred direction of this transceiver. An update of 98 // directionality does not take effect immediately. Instead, future calls to 99 // CreateOffer and CreateAnswer mark the corresponding media descriptions as 100 // sendrecv, sendonly, recvonly, or inactive. 101 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction 102 virtual void SetDirection(RtpTransceiverDirection new_direction) = 0; 103 104 // The current_direction attribute indicates the current direction negotiated 105 // for this transceiver. If this transceiver has never been represented in an 106 // offer/answer exchange, or if the transceiver is stopped, the value is null. 107 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection 108 virtual absl::optional<RtpTransceiverDirection> current_direction() const = 0; 109 110 // An internal slot designating for which direction the relevant 111 // PeerConnection events have been fired. This is to ensure that events like 112 // OnAddTrack only get fired once even if the same session description is 113 // applied again. 114 // Exposed in the public interface for use by Chromium. 115 virtual absl::optional<RtpTransceiverDirection> fired_direction() const; 116 117 // The Stop method irreversibly stops the RtpTransceiver. The sender of this 118 // transceiver will no longer send, the receiver will no longer receive. 119 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop 120 virtual void Stop() = 0; 121 122 // The SetCodecPreferences method overrides the default codec preferences used 123 // by WebRTC for this transceiver. 124 // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-setcodecpreferences 125 virtual RTCError SetCodecPreferences( 126 rtc::ArrayView<RtpCodecCapability> codecs); 127 virtual std::vector<RtpCodecCapability> codec_preferences() const; 128 129 // Readonly attribute which contains the set of header extensions that was set 130 // with SetOfferedRtpHeaderExtensions, or a default set if it has not been 131 // called. 132 // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface 133 virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer() 134 const; 135 136 // The SetOfferedRtpHeaderExtensions method modifies the next SDP negotiation 137 // so that it negotiates use of header extensions which are not kStopped. 138 // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface 139 virtual webrtc::RTCError SetOfferedRtpHeaderExtensions( 140 rtc::ArrayView<const RtpHeaderExtensionCapability> 141 header_extensions_to_offer); 142 143 protected: 144 ~RtpTransceiverInterface() override = default; 145 }; 146 147 } // namespace webrtc 148 149 #endif // API_RTP_TRANSCEIVER_INTERFACE_H_ 150