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/rtp_parameters.h" 26 #include "api/scoped_refptr.h" 27 #include "api/transport/rtp/rtp_source.h" 28 #include "rtc_base/ref_count.h" 29 #include "rtc_base/system/rtc_export.h" 30 31 namespace webrtc { 32 33 class RtpReceiverObserverInterface { 34 public: 35 // Note: Currently if there are multiple RtpReceivers of the same media type, 36 // they will all call OnFirstPacketReceived at once. 37 // 38 // In the future, it's likely that an RtpReceiver will only call 39 // OnFirstPacketReceived when a packet is received specifically for its 40 // SSRC/mid. 41 virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; 42 43 protected: ~RtpReceiverObserverInterface()44 virtual ~RtpReceiverObserverInterface() {} 45 }; 46 47 class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface { 48 public: 49 virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; 50 51 // The dtlsTransport attribute exposes the DTLS transport on which the 52 // media is received. It may be null. 53 // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport 54 // TODO(https://bugs.webrtc.org/907849) remove default implementation 55 virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const; 56 57 // The list of streams that `track` is associated with. This is the same as 58 // the [[AssociatedRemoteMediaStreams]] internal slot in the spec. 59 // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams 60 // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this. 61 // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of 62 // stream_ids() as soon as downstream projects are no longer dependent on 63 // stream objects. 64 virtual std::vector<std::string> stream_ids() const; 65 virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const; 66 67 // Audio or video receiver? 68 virtual cricket::MediaType media_type() const = 0; 69 70 // Not to be confused with "mid", this is a field we can temporarily use 71 // to uniquely identify a receiver until we implement Unified Plan SDP. 72 virtual std::string id() const = 0; 73 74 // The WebRTC specification only defines RTCRtpParameters in terms of senders, 75 // but this API also applies them to receivers, similar to ORTC: 76 // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. 77 virtual RtpParameters GetParameters() const = 0; 78 // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium. 79 // Currently, doesn't support changing any parameters. SetParameters(const RtpParameters & parameters)80 virtual bool SetParameters(const RtpParameters& parameters) { return false; } 81 82 // Does not take ownership of observer. 83 // Must call SetObserver(nullptr) before the observer is destroyed. 84 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; 85 86 // Sets the jitter buffer minimum delay until media playout. Actual observed 87 // delay may differ depending on the congestion control. `delay_seconds` is a 88 // positive value including 0.0 measured in seconds. `nullopt` means default 89 // value must be used. 90 virtual void SetJitterBufferMinimumDelay( 91 absl::optional<double> delay_seconds) = 0; 92 93 // TODO(zhihuang): Remove the default implementation once the subclasses 94 // implement this. Currently, the only relevant subclass is the 95 // content::FakeRtpReceiver in Chromium. 96 virtual std::vector<RtpSource> GetSources() const; 97 98 // Sets a user defined frame decryptor that will decrypt the entire frame 99 // before it is sent across the network. This will decrypt the entire frame 100 // using the user provided decryption mechanism regardless of whether SRTP is 101 // enabled or not. 102 // TODO(bugs.webrtc.org/12772): Remove. 103 virtual void SetFrameDecryptor( 104 rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor); 105 106 // Returns a pointer to the frame decryptor set previously by the 107 // user. This can be used to update the state of the object. 108 // TODO(bugs.webrtc.org/12772): Remove. 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 } // namespace webrtc 122 123 #endif // API_RTP_RECEIVER_INTERFACE_H_ 124