/* * 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 #include #include #include "absl/types/optional.h" #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/rtp_parameters.h" #include "api/rtp_receiver_interface.h" #include "api/scoped_refptr.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "media/base/media_channel.h" #include "media/base/video_broadcaster.h" #include "pc/video_track_source.h" #include "rtc_base/ref_counted_object.h" #include "rtc_base/thread.h" namespace webrtc { // Internal class used by PeerConnection. class RtpReceiverInternal : public RtpReceiverInterface { public: // Stops receiving. The track may be reactivated. virtual void Stop() = 0; // Stops the receiver permanently. // Causes the associated track to enter kEnded state. Cannot be reversed. virtual void StopAndEndTrack() = 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. // Must call SetMediaChannel(nullptr) before the media channel is destroyed. virtual void SetMediaChannel(cricket::MediaChannel* 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 dtls_transport) = 0; // This SSRC is used as an identifier for the receiver between the API layer // and the WebRtcVideoEngine, WebRtcVoiceEngine layer. virtual 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 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>& 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> CreateStreamsFromIds(std::vector stream_ids); static void MaybeAttachFrameDecryptorToMediaChannel( const absl::optional& ssrc, rtc::Thread* worker_thread, rtc::scoped_refptr frame_decryptor, cricket::MediaChannel* media_channel, bool stopped); }; } // namespace webrtc #endif // PC_RTP_RECEIVER_H_