| /* |
| * 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 RtpReceivers |
| // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface |
| |
| #ifndef API_RTP_RECEIVER_INTERFACE_H_ |
| #define API_RTP_RECEIVER_INTERFACE_H_ |
| |
| #include <string> |
| #include <vector> |
| |
| #include "api/crypto/frame_decryptor_interface.h" |
| #include "api/dtls_transport_interface.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/proxy.h" |
| #include "api/rtp_parameters.h" |
| #include "api/scoped_refptr.h" |
| #include "api/transport/rtp/rtp_source.h" |
| #include "rtc_base/deprecation.h" |
| #include "rtc_base/ref_count.h" |
| |
| namespace webrtc { |
| |
| class RtpReceiverObserverInterface { |
| public: |
| // Note: Currently if there are multiple RtpReceivers of the same media type, |
| // they will all call OnFirstPacketReceived at once. |
| // |
| // In the future, it's likely that an RtpReceiver will only call |
| // OnFirstPacketReceived when a packet is received specifically for its |
| // SSRC/mid. |
| virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0; |
| |
| protected: |
| virtual ~RtpReceiverObserverInterface() {} |
| }; |
| |
| class RtpReceiverInterface : public rtc::RefCountInterface { |
| public: |
| virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; |
| |
| // The dtlsTransport attribute exposes the DTLS transport on which the |
| // media is received. It may be null. |
| // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport |
| // TODO(https://bugs.webrtc.org/907849) remove default implementation |
| virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const; |
| |
| // The list of streams that |track| is associated with. This is the same as |
| // the [[AssociatedRemoteMediaStreams]] internal slot in the spec. |
| // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams |
| // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this. |
| // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of |
| // stream_ids() as soon as downstream projects are no longer dependent on |
| // stream objects. |
| virtual std::vector<std::string> stream_ids() const; |
| virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const; |
| |
| // Audio or video receiver? |
| 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; |
| |
| // The WebRTC specification only defines RTCRtpParameters in terms of senders, |
| // but this API also applies them to receivers, similar to ORTC: |
| // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*. |
| virtual RtpParameters GetParameters() const = 0; |
| // Currently, doesn't support changing any parameters, but may in the future. |
| virtual bool SetParameters(const RtpParameters& parameters) = 0; |
| |
| // Does not take ownership of observer. |
| // Must call SetObserver(nullptr) before the observer is destroyed. |
| virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0; |
| |
| // Sets the jitter buffer minimum delay until media playout. Actual observed |
| // delay may differ depending on the congestion control. |delay_seconds| is a |
| // positive value including 0.0 measured in seconds. |nullopt| means default |
| // value must be used. |
| virtual void SetJitterBufferMinimumDelay( |
| absl::optional<double> delay_seconds) = 0; |
| |
| // TODO(zhihuang): Remove the default implementation once the subclasses |
| // implement this. Currently, the only relevant subclass is the |
| // content::FakeRtpReceiver in Chromium. |
| virtual std::vector<RtpSource> GetSources() const; |
| |
| // Sets a user defined frame decryptor that will decrypt the entire frame |
| // before it is sent across the network. This will decrypt the entire frame |
| // using the user provided decryption mechanism regardless of whether SRTP is |
| // enabled or not. |
| virtual void SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor); |
| |
| // Returns a pointer to the frame decryptor set previously by the |
| // user. This can be used to update the state of the object. |
| virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const; |
| |
| protected: |
| ~RtpReceiverInterface() override = default; |
| }; |
| |
| // Define proxy for RtpReceiverInterface. |
| // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods |
| // are called on is an implementation detail. |
| BEGIN_SIGNALING_PROXY_MAP(RtpReceiver) |
| PROXY_SIGNALING_THREAD_DESTRUCTOR() |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport) |
| PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids) |
| PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>, |
| streams) |
| PROXY_CONSTMETHOD0(cricket::MediaType, media_type) |
| PROXY_CONSTMETHOD0(std::string, id) |
| PROXY_CONSTMETHOD0(RtpParameters, GetParameters) |
| PROXY_METHOD1(bool, SetParameters, const RtpParameters&) |
| PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*) |
| PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>) |
| PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources) |
| PROXY_METHOD1(void, |
| SetFrameDecryptor, |
| rtc::scoped_refptr<FrameDecryptorInterface>) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>, |
| GetFrameDecryptor) |
| END_PROXY_MAP() |
| |
| } // namespace webrtc |
| |
| #endif // API_RTP_RECEIVER_INTERFACE_H_ |