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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/rtp_transport.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/rtp_transport.h')
-rw-r--r-- | third_party/libwebrtc/pc/rtp_transport.h | 144 |
1 files changed, 144 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/rtp_transport.h b/third_party/libwebrtc/pc/rtp_transport.h new file mode 100644 index 0000000000..1afcb5ee3d --- /dev/null +++ b/third_party/libwebrtc/pc/rtp_transport.h @@ -0,0 +1,144 @@ +/* + * 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 <stddef.h> +#include <stdint.h> + +#include <string> + +#include "absl/types/optional.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); + + // Overridden by SrtpTransport. + virtual void OnNetworkRouteChanged( + absl::optional<rtc::NetworkRoute> 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_; +}; + +} // namespace webrtc + +#endif // PC_RTP_TRANSPORT_H_ |