| /* |
| * 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 cricket::VoiceChannel/cricket::VideoChannel) |
| |
| #ifndef PC_RTP_RECEIVER_H_ |
| #define PC_RTP_RECEIVER_H_ |
| |
| #include <stdint.h> |
| |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.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( |
| cricket::MediaReceiveChannelInterface* media_channel) = 0; |
| |
| // Configures the RtpReceiver with the underlying media channel, with the |
| // given SSRC as the stream identifier. |
| virtual void SetupMediaChannel(uint32_t ssrc) = 0; |
| |
| // Configures the RtpReceiver with the underlying media channel to receive an |
| // unsignaled receive stream. |
| virtual void SetupUnsignaledMediaChannel() = 0; |
| |
| virtual void set_transport( |
| rtc::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 absl::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() = 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<rtc::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<rtc::scoped_refptr<MediaStreamInterface>> |
| CreateStreamsFromIds(std::vector<std::string> stream_ids); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_RTP_RECEIVER_H_ |