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 RtpReceivers 12 // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface 13 14 #ifndef API_RTP_RECEIVER_INTERFACE_H_ 15 #define API_RTP_RECEIVER_INTERFACE_H_ 16 17 #include <string> 18 #include <vector> 19 20 #include "api/crypto/frame_decryptor_interface.h" 21 #include "api/dtls_transport_interface.h" 22 #include "api/frame_transformer_interface.h" 23 #include "api/media_stream_interface.h" 24 #include "api/media_types.h" 25 #include "api/proxy.h" 26 #include "api/rtp_parameters.h" 27 #include "api/scoped_refptr.h" 28 #include "api/transport/rtp/rtp_source.h" 29 #include "rtc_base/deprecation.h" 30 #include "rtc_base/ref_count.h" 31 #include "rtc_base/system/rtc_export.h" 32 33 namespace webrtc { 34 35 class RtpReceiverObserverInterface { 36 public: 37 // Note: Currently if there are multiple RtpReceivers of the same media type, 38 // they will all call OnFirstPacketReceived at once. 39 // 40 // In the future, it's likely that an RtpReceiver will only call 41 // OnFirstPacketReceived when a packet is received specifically for its 42 // SSRC/mid. 43 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; 44 45 protected: ~RtpReceiverObserverInterface()46 virtual ~RtpReceiverObserverInterface() {} 47 }; 48 49 class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface { 50 public: 51 virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; 52 53 // The dtlsTransport attribute exposes the DTLS transport on which the 54 // media is received. It may be null. 55 // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport 56 // TODO(https://bugs.webrtc.org/907849) remove default implementation 57 virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const; 58 59 // The list of streams that |track| is associated with. This is the same as 60 // the [[AssociatedRemoteMediaStreams]] internal slot in the spec. 61 // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams 62 // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this. 63 // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of 64 // stream_ids() as soon as downstream projects are no longer dependent on 65 // stream objects. 66 virtual std::vector<std::string> stream_ids() const; 67 virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const; 68 69 // Audio or video receiver? 70 virtual cricket::MediaType media_type() const = 0; 71 72 // Not to be confused with "mid", this is a field we can temporarily use 73 // to uniquely identify a receiver until we implement Unified Plan SDP. 74 virtual std::string id() const = 0; 75 76 // The WebRTC specification only defines RTCRtpParameters in terms of senders, 77 // but this API also applies them to receivers, similar to ORTC: 78 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. 79 virtual RtpParameters GetParameters() const = 0; 80 // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium. 81 // Currently, doesn't support changing any parameters. SetParameters(const RtpParameters & parameters)82 virtual bool SetParameters(const RtpParameters& parameters) { return false; } 83 84 // Does not take ownership of observer. 85 // Must call SetObserver(nullptr) before the observer is destroyed. 86 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; 87 88 // Sets the jitter buffer minimum delay until media playout. Actual observed 89 // delay may differ depending on the congestion control. |delay_seconds| is a 90 // positive value including 0.0 measured in seconds. |nullopt| means default 91 // value must be used. 92 virtual void SetJitterBufferMinimumDelay( 93 absl::optional<double> delay_seconds) = 0; 94 95 // TODO(zhihuang): Remove the default implementation once the subclasses 96 // implement this. Currently, the only relevant subclass is the 97 // content::FakeRtpReceiver in Chromium. 98 virtual std::vector<RtpSource> GetSources() const; 99 100 // Sets a user defined frame decryptor that will decrypt the entire frame 101 // before it is sent across the network. This will decrypt the entire frame 102 // using the user provided decryption mechanism regardless of whether SRTP is 103 // enabled or not. 104 virtual void SetFrameDecryptor( 105 rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor); 106 107 // Returns a pointer to the frame decryptor set previously by the 108 // user. This can be used to update the state of the object. 109 virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const; 110 111 // Sets a frame transformer between the depacketizer and the decoder to enable 112 // client code to transform received frames according to their own processing 113 // logic. 114 virtual void SetDepacketizerToDecoderFrameTransformer( 115 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer); 116 117 protected: 118 ~RtpReceiverInterface() override = default; 119 }; 120 121 // Define proxy for RtpReceiverInterface. 122 // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods 123 // are called on is an implementation detail. 124 BEGIN_SIGNALING_PROXY_MAP(RtpReceiver) 125 PROXY_SIGNALING_THREAD_DESTRUCTOR() 126 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) 127 PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport) 128 PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids) 129 PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>, 130 streams) 131 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) 132 PROXY_CONSTMETHOD0(std::string, id) 133 PROXY_CONSTMETHOD0(RtpParameters, GetParameters) 134 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*) 135 PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>) 136 PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources) 137 PROXY_METHOD1(void, 138 SetFrameDecryptor, 139 rtc::scoped_refptr<FrameDecryptorInterface>) 140 PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>, 141 GetFrameDecryptor) 142 PROXY_METHOD1(void, 143 SetDepacketizerToDecoderFrameTransformer, 144 rtc::scoped_refptr<FrameTransformerInterface>) 145 END_PROXY_MAP() 146 147 } // namespace webrtc 148 149 #endif // API_RTP_RECEIVER_INTERFACE_H_ 150