| /* | |
| * 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 classes that implement RtpReceiverInterface. | |
| // An RtpReceiver associates a MediaStreamTrackInterface with an underlying | |
| // transport (provided by webrtc::VoiceChannel/webrtc::VideoChannel) | |
| #ifndef PC_RTP_RECEIVER_H_ | |
| #define PC_RTP_RECEIVER_H_ | |
| #include <stdint.h> | |
| #include <optional> | |
| #include <string> | |
| #include <vector> | |
| #include "absl/functional/any_invocable.h" | |
| #include "api/dtls_transport_interface.h" | |
| #include "api/media_stream_interface.h" | |
| #include "api/rtp_receiver_interface.h" | |
| #include "api/scoped_refptr.h" | |
| #include "media/base/media_channel.h" | |
| namespace webrtc { | |
| // Internal class used by PeerConnection. | |
| class RtpReceiverInternal : public RtpReceiverInterface { | |
| public: | |
| // Call on the signaling thread, to let the receiver know that the the | |
| // embedded source object should enter a stopped/ended state and the track's | |
| // state set to `kEnded`, a final state that cannot be reversed. | |
| virtual void Stop() = 0; | |
| // Sets the underlying MediaEngine channel associated with this RtpSender. | |
| // A VoiceMediaChannel should be used for audio RtpSenders and | |
| // a VideoMediaChannel should be used for video RtpSenders. | |
| // NOTE: | |
| // * SetMediaChannel(nullptr) must be called before the media channel is | |
| // destroyed. | |
| // * This method must be invoked on the worker thread. | |
| virtual void SetMediaChannel(MediaReceiveChannelInterface* media_channel) = 0; | |
| // Configures the RtpReceiver with the underlying media channel, with the | |
| // given SSRC as the stream identifier. | |
| // This function returns a callable object that must be invoked on the worker | |
| // thread by the caller to complete the operation. This is done to allow the | |
| // caller to batch up tasks for the worker thread. | |
| [[nodiscard]] virtual absl::AnyInvocable<void() &&> GetSetupForMediaChannel( | |
| uint32_t ssrc) = 0; | |
| // Configures the RtpReceiver with the underlying media channel to receive an | |
| // unsignaled receive stream. | |
| // This function returns a callable object that must be invoked on the worker | |
| // thread by the caller to complete the operation. This is done to allow the | |
| // caller to batch up tasks for the worker thread. | |
| [[nodiscard]] virtual absl::AnyInvocable<void() &&> | |
| GetSetupForUnsignaledMediaChannel() = 0; | |
| virtual void set_transport( | |
| scoped_refptr<DtlsTransportInterface> dtls_transport) = 0; | |
| // This SSRC is used as an identifier for the receiver between the API layer | |
| // and the WebRtcVideoEngine, WebRtcVoiceEngine layer. | |
| virtual std::optional<uint32_t> ssrc() const = 0; | |
| // Call this to notify the RtpReceiver when the first packet has been received | |
| // on the corresponding channel. | |
| virtual void NotifyFirstPacketReceived(uint32_t ssrc) = 0; | |
| // Similar to the above but done whenever the receptive flag changed. | |
| virtual void NotifyFirstPacketReceivedAfterReceptiveChange(uint32_t ssrc) = 0; | |
| // Set the associated remote media streams for this receiver. The remote track | |
| // will be removed from any streams that are no longer present and added to | |
| // any new streams. | |
| virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0; | |
| // TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of | |
| // set_stream_ids() as soon as downstream projects are no longer dependent on | |
| // stream objects. | |
| virtual void SetStreams( | |
| const std::vector<scoped_refptr<MediaStreamInterface>>& streams) = 0; | |
| // Returns an ID that changes if the attached track changes, but | |
| // otherwise remains constant. Used to generate IDs for stats. | |
| // The special value zero means that no track is attached. | |
| virtual int AttachmentId() const = 0; | |
| protected: | |
| static int GenerateUniqueId(); | |
| static std::vector<scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds( | |
| std::vector<std::string> stream_ids); | |
| }; | |
| } // namespace webrtc | |
| #endif // PC_RTP_RECEIVER_H_ |