| /* |
| * 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 RtpSenders |
| // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface |
| |
| #ifndef API_RTP_SENDER_INTERFACE_H_ |
| #define API_RTP_SENDER_INTERFACE_H_ |
| |
| #include <string> |
| #include <vector> |
| |
| #include "api/crypto/frame_encryptor_interface.h" |
| #include "api/dtmf_sender_interface.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/proxy.h" |
| #include "api/rtc_error.h" |
| #include "api/rtp_parameters.h" |
| #include "rtc_base/ref_count.h" |
| #include "rtc_base/scoped_ref_ptr.h" |
| |
| namespace webrtc { |
| |
| class RtpSenderInterface : public rtc::RefCountInterface { |
| public: |
| // Returns true if successful in setting the track. |
| // Fails if an audio track is set on a video RtpSender, or vice-versa. |
| virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; |
| virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0; |
| |
| // Returns primary SSRC used by this sender for sending media. |
| // Returns 0 if not yet determined. |
| // TODO(deadbeef): Change to absl::optional. |
| // TODO(deadbeef): Remove? With GetParameters this should be redundant. |
| virtual uint32_t ssrc() const = 0; |
| |
| // Audio or video sender? |
| 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; |
| |
| // Returns a list of media stream ids associated with this sender's track. |
| // These are signalled in the SDP so that the remote side can associate |
| // tracks. |
| virtual std::vector<std::string> stream_ids() const = 0; |
| |
| // Returns the list of encoding parameters that will be applied when the SDP |
| // local description is set. These initial encoding parameters can be set by |
| // PeerConnection::AddTransceiver, and later updated with Get/SetParameters. |
| // TODO(orphis): Make it pure virtual once Chrome has updated |
| virtual std::vector<RtpEncodingParameters> init_send_encodings() const; |
| |
| virtual RtpParameters GetParameters() = 0; |
| // Note that only a subset of the parameters can currently be changed. See |
| // rtpparameters.h |
| // The encodings are in increasing quality order for simulcast. |
| virtual RTCError SetParameters(const RtpParameters& parameters) = 0; |
| |
| // Returns null for a video sender. |
| virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0; |
| |
| // Sets a user defined frame encryptor that will encrypt the entire frame |
| // before it is sent across the network. This will encrypt the entire frame |
| // using the user provided encryption mechanism regardless of whether SRTP is |
| // enabled or not. |
| virtual void SetFrameEncryptor( |
| rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor); |
| |
| // Returns a pointer to the frame encryptor set previously by the |
| // user. This can be used to update the state of the object. |
| virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const; |
| |
| protected: |
| ~RtpSenderInterface() override = default; |
| }; |
| |
| // Define proxy for RtpSenderInterface. |
| // 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(RtpSender) |
| PROXY_SIGNALING_THREAD_DESTRUCTOR() |
| PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track) |
| PROXY_CONSTMETHOD0(uint32_t, ssrc) |
| PROXY_CONSTMETHOD0(cricket::MediaType, media_type) |
| PROXY_CONSTMETHOD0(std::string, id) |
| PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids) |
| PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings) |
| PROXY_METHOD0(RtpParameters, GetParameters); |
| PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&) |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender); |
| PROXY_METHOD1(void, |
| SetFrameEncryptor, |
| rtc::scoped_refptr<FrameEncryptorInterface>); |
| PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>, |
| GetFrameEncryptor); |
| END_PROXY_MAP() |
| |
| } // namespace webrtc |
| |
| #endif // API_RTP_SENDER_INTERFACE_H_ |