3a1b92772f
RtpTransportInternal does not need to expose these. They are only used by tests and for setting options. Instead, it can expose a SetRtpOption and SetRtcpOption to set options relevant to each of its transports. Also updates tests to work around no longer having access to internals. This will simplify the composite needed during negotiation of different RTP transport types, as we no longer need to have composites of both RtpTransport and PacketTransport. Bug: webrtc:9719 Change-Id: I91bfa6e95b7aa384d10497f47e7d2483c2e0bef2 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138282 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28066}
134 lines
4.3 KiB
C++
134 lines
4.3 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 <string>
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#include "call/rtp_demuxer.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "pc/rtp_transport_internal.h"
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#include "rtc_base/third_party/sigslot/sigslot.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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explicit RtpTransport(bool rtcp_mux_enabled)
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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, int64_t packet_time_us);
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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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// Overridden by SrtpTransport.
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virtual void OnNetworkRouteChanged(
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absl::optional<rtc::NetworkRoute> network_route);
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virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us);
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virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
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int64_t packet_time_us);
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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 char* data,
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size_t len,
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const int64_t& packet_time_us,
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int flags);
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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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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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};
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} // namespace webrtc
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#endif // PC_RTP_TRANSPORT_H_
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