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8037fc6ffa
Bug: webrtc:342905193 No-Try: True Change-Id: Icc968be43b8830038ea9a1f5f604307220457807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021 Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42911}
153 lines
5 KiB
C++
153 lines
5 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTP_TRANSPORT_H_
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#define PC_RTP_TRANSPORT_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <optional>
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#include <string>
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#include "api/field_trials_view.h"
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#include "api/task_queue/pending_task_safety_flag.h"
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#include "api/units/timestamp.h"
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#include "call/rtp_demuxer.h"
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#include "call/video_receive_stream.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "p2p/base/packet_transport_internal.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/session_description.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/received_packet.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/socket.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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class PacketTransportInternal;
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} // namespace rtc
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namespace webrtc {
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class RtpTransport : public RtpTransportInternal {
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public:
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RtpTransport(const RtpTransport&) = delete;
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RtpTransport& operator=(const RtpTransport&) = delete;
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RtpTransport(bool rtcp_mux_enabled, const FieldTrialsView& field_trials)
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: set_ready_to_send_false_if_send_fail_(
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field_trials.IsEnabled("WebRTC-SetReadyToSendFalseIfSendFail")),
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rtcp_mux_enabled_(rtcp_mux_enabled) {}
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bool rtcp_mux_enabled() const override { return rtcp_mux_enabled_; }
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void SetRtcpMuxEnabled(bool enable) override;
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const std::string& transport_name() const override;
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int SetRtpOption(rtc::Socket::Option opt, int value) override;
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int SetRtcpOption(rtc::Socket::Option opt, int value) override;
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rtc::PacketTransportInternal* rtp_packet_transport() const {
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return rtp_packet_transport_;
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}
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void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp);
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rtc::PacketTransportInternal* rtcp_packet_transport() const {
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return rtcp_packet_transport_;
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}
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void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp);
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bool IsReadyToSend() const override { return ready_to_send_; }
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bool IsWritable(bool rtcp) const override;
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bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool IsSrtpActive() const override { return false; }
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void UpdateRtpHeaderExtensionMap(
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const cricket::RtpHeaderExtensions& header_extensions) override;
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bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
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RtpPacketSinkInterface* sink) override;
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bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
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protected:
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// These methods will be used in the subclasses.
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void DemuxPacket(rtc::CopyOnWriteBuffer packet,
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Timestamp arrival_time,
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rtc::EcnMarking ecn);
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bool SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags);
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flat_set<uint32_t> GetSsrcsForSink(RtpPacketSinkInterface* sink);
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// Overridden by SrtpTransport.
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virtual void OnNetworkRouteChanged(
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std::optional<rtc::NetworkRoute> network_route);
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virtual void OnRtpPacketReceived(const rtc::ReceivedPacket& packet);
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virtual void OnRtcpPacketReceived(const rtc::ReceivedPacket& packet);
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// Overridden by SrtpTransport and DtlsSrtpTransport.
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virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport);
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private:
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void OnReadyToSend(rtc::PacketTransportInternal* transport);
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void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
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const rtc::SentPacket& sent_packet);
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void OnReadPacket(rtc::PacketTransportInternal* transport,
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const rtc::ReceivedPacket& received_packet);
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// Updates "ready to send" for an individual channel and fires
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// SignalReadyToSend.
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void SetReadyToSend(bool rtcp, bool ready);
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void MaybeSignalReadyToSend();
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bool IsTransportWritable();
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const bool set_ready_to_send_false_if_send_fail_;
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bool rtcp_mux_enabled_;
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rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
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rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
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bool ready_to_send_ = false;
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bool rtp_ready_to_send_ = false;
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bool rtcp_ready_to_send_ = false;
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RtpDemuxer rtp_demuxer_;
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// Used for identifying the MID for RtpDemuxer.
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RtpHeaderExtensionMap header_extension_map_;
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// Guard against recursive "ready to send" signals
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bool processing_ready_to_send_ = false;
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bool processing_sent_packet_ = false;
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ScopedTaskSafety safety_;
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};
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} // namespace webrtc
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#endif // PC_RTP_TRANSPORT_H_
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