/* * Copyright 2017 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. */ #ifndef PC_RTP_TRANSPORT_H_ #define PC_RTP_TRANSPORT_H_ #include #include #include #include "absl/types/optional.h" #include "api/task_queue/pending_task_safety_flag.h" #include "call/rtp_demuxer.h" #include "call/video_receive_stream.h" #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" #include "p2p/base/packet_transport_internal.h" #include "pc/rtp_transport_internal.h" #include "pc/session_description.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/network/sent_packet.h" #include "rtc_base/network_route.h" #include "rtc_base/socket.h" namespace rtc { class CopyOnWriteBuffer; struct PacketOptions; class PacketTransportInternal; } // namespace rtc namespace webrtc { class RtpTransport : public RtpTransportInternal { public: RtpTransport(const RtpTransport&) = delete; RtpTransport& operator=(const RtpTransport&) = delete; explicit RtpTransport(bool rtcp_mux_enabled) : rtcp_mux_enabled_(rtcp_mux_enabled) {} bool rtcp_mux_enabled() const override { return rtcp_mux_enabled_; } void SetRtcpMuxEnabled(bool enable) override; const std::string& transport_name() const override; int SetRtpOption(rtc::Socket::Option opt, int value) override; int SetRtcpOption(rtc::Socket::Option opt, int value) override; rtc::PacketTransportInternal* rtp_packet_transport() const { return rtp_packet_transport_; } void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp); rtc::PacketTransportInternal* rtcp_packet_transport() const { return rtcp_packet_transport_; } void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp); bool IsReadyToSend() const override { return ready_to_send_; } bool IsWritable(bool rtcp) const override; bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options, int flags) override; bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options, int flags) override; bool IsSrtpActive() const override { return false; } void UpdateRtpHeaderExtensionMap( const cricket::RtpHeaderExtensions& header_extensions) override; bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria, RtpPacketSinkInterface* sink) override; bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override; protected: // These methods will be used in the subclasses. void DemuxPacket(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us); bool SendPacket(bool rtcp, rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options, int flags); flat_set GetSsrcsForSink(RtpPacketSinkInterface* sink); // Overridden by SrtpTransport. virtual void OnNetworkRouteChanged( absl::optional network_route); virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us); virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us); // Overridden by SrtpTransport and DtlsSrtpTransport. virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport); private: void OnReadyToSend(rtc::PacketTransportInternal* transport); void OnSentPacket(rtc::PacketTransportInternal* packet_transport, const rtc::SentPacket& sent_packet); void OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t len, const int64_t& packet_time_us, int flags); // Updates "ready to send" for an individual channel and fires // SignalReadyToSend. void SetReadyToSend(bool rtcp, bool ready); void MaybeSignalReadyToSend(); bool IsTransportWritable(); bool rtcp_mux_enabled_; rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr; rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr; bool ready_to_send_ = false; bool rtp_ready_to_send_ = false; bool rtcp_ready_to_send_ = false; RtpDemuxer rtp_demuxer_; // Used for identifying the MID for RtpDemuxer. RtpHeaderExtensionMap header_extension_map_; // Guard against recursive "ready to send" signals bool processing_ready_to_send_ = false; bool processing_sent_packet_ = false; ScopedTaskSafety safety_; }; } // namespace webrtc #endif // PC_RTP_TRANSPORT_H_