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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/pc/sctp_data_channel.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/sctp_data_channel.h')
-rw-r--r-- | third_party/libwebrtc/pc/sctp_data_channel.h | 295 |
1 files changed, 295 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/sctp_data_channel.h b/third_party/libwebrtc/pc/sctp_data_channel.h new file mode 100644 index 0000000000..a8442c59cc --- /dev/null +++ b/third_party/libwebrtc/pc/sctp_data_channel.h @@ -0,0 +1,295 @@ +/* + * Copyright 2020 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_SCTP_DATA_CHANNEL_H_ +#define PC_SCTP_DATA_CHANNEL_H_ + +#include <stdint.h> + +#include <memory> +#include <set> +#include <string> + +#include "absl/types/optional.h" +#include "api/data_channel_interface.h" +#include "api/priority.h" +#include "api/rtc_error.h" +#include "api/scoped_refptr.h" +#include "api/transport/data_channel_transport_interface.h" +#include "media/base/media_channel.h" +#include "pc/data_channel_utils.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/ssl_stream_adapter.h" // For SSLRole +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" + +namespace webrtc { + +class SctpDataChannel; + +// TODO(deadbeef): Get rid of this and have SctpDataChannel depend on +// SctpTransportInternal (pure virtual SctpTransport interface) instead. +class SctpDataChannelControllerInterface { + public: + // Sends the data to the transport. + virtual bool SendData(int sid, + const SendDataParams& params, + const rtc::CopyOnWriteBuffer& payload, + cricket::SendDataResult* result) = 0; + // Connects to the transport signals. + virtual bool ConnectDataChannel(SctpDataChannel* data_channel) = 0; + // Disconnects from the transport signals. + virtual void DisconnectDataChannel(SctpDataChannel* data_channel) = 0; + // Adds the data channel SID to the transport for SCTP. + virtual void AddSctpDataStream(int sid) = 0; + // Begins the closing procedure by sending an outgoing stream reset. Still + // need to wait for callbacks to tell when this completes. + virtual void RemoveSctpDataStream(int sid) = 0; + // Returns true if the transport channel is ready to send data. + virtual bool ReadyToSendData() const = 0; + + protected: + virtual ~SctpDataChannelControllerInterface() {} +}; + +// TODO(tommi): Change to not inherit from DataChannelInit but to have it as +// a const member. Block access to the 'id' member since it cannot be const. +struct InternalDataChannelInit : public DataChannelInit { + enum OpenHandshakeRole { kOpener, kAcker, kNone }; + // The default role is kOpener because the default `negotiated` is false. + InternalDataChannelInit() : open_handshake_role(kOpener) {} + explicit InternalDataChannelInit(const DataChannelInit& base); + OpenHandshakeRole open_handshake_role; +}; + +// Helper class to allocate unique IDs for SCTP DataChannels. +class SctpSidAllocator { + public: + // Gets the first unused odd/even id based on the DTLS role. If `role` is + // SSL_CLIENT, the allocated id starts from 0 and takes even numbers; + // otherwise, the id starts from 1 and takes odd numbers. + // Returns false if no ID can be allocated. + bool AllocateSid(rtc::SSLRole role, int* sid); + + // Attempts to reserve a specific sid. Returns false if it's unavailable. + bool ReserveSid(int sid); + + // Indicates that `sid` isn't in use any more, and is thus available again. + void ReleaseSid(int sid); + + private: + // Checks if `sid` is available to be assigned to a new SCTP data channel. + bool IsSidAvailable(int sid) const; + + std::set<int> used_sids_; +}; + +// SctpDataChannel is an implementation of the DataChannelInterface based on +// SctpTransport. It provides an implementation of unreliable or +// reliabledata channels. + +// DataChannel states: +// kConnecting: The channel has been created the transport might not yet be +// ready. +// kOpen: The open handshake has been performed (if relevant) and the data +// channel is able to send messages. +// kClosing: DataChannelInterface::Close has been called, or the remote side +// initiated the closing procedure, but the closing procedure has not +// yet finished. +// kClosed: The closing handshake is finished (possibly initiated from this, +// side, possibly from the peer). +// +// How the closing procedure works for SCTP: +// 1. Alice calls Close(), state changes to kClosing. +// 2. Alice finishes sending any queued data. +// 3. Alice calls RemoveSctpDataStream, sends outgoing stream reset. +// 4. Bob receives incoming stream reset; OnClosingProcedureStartedRemotely +// called. +// 5. Bob sends outgoing stream reset. +// 6. Alice receives incoming reset, Bob receives acknowledgement. Both receive +// OnClosingProcedureComplete callback and transition to kClosed. +class SctpDataChannel : public DataChannelInterface, + public sigslot::has_slots<> { + public: + static rtc::scoped_refptr<SctpDataChannel> Create( + SctpDataChannelControllerInterface* controller, + const std::string& label, + const InternalDataChannelInit& config, + rtc::Thread* signaling_thread, + rtc::Thread* network_thread); + + // Instantiates an API proxy for a SctpDataChannel instance that will be + // handed out to external callers. + static rtc::scoped_refptr<DataChannelInterface> CreateProxy( + rtc::scoped_refptr<SctpDataChannel> channel); + + // Invalidate the link to the controller (DataChannelController); + void DetachFromController(); + + void RegisterObserver(DataChannelObserver* observer) override; + void UnregisterObserver() override; + + std::string label() const override { return label_; } + bool reliable() const override; + bool ordered() const override { return config_.ordered; } + // Backwards compatible accessors + uint16_t maxRetransmitTime() const override { + return config_.maxRetransmitTime ? *config_.maxRetransmitTime + : static_cast<uint16_t>(-1); + } + uint16_t maxRetransmits() const override { + return config_.maxRetransmits ? *config_.maxRetransmits + : static_cast<uint16_t>(-1); + } + absl::optional<int> maxPacketLifeTime() const override { + return config_.maxRetransmitTime; + } + absl::optional<int> maxRetransmitsOpt() const override { + return config_.maxRetransmits; + } + std::string protocol() const override { return config_.protocol; } + bool negotiated() const override { return config_.negotiated; } + int id() const override { return config_.id; } + Priority priority() const override { + return config_.priority ? *config_.priority : Priority::kLow; + } + + virtual int internal_id() const { return internal_id_; } + + uint64_t buffered_amount() const override; + void Close() override; + DataState state() const override; + RTCError error() const override; + uint32_t messages_sent() const override; + uint64_t bytes_sent() const override; + uint32_t messages_received() const override; + uint64_t bytes_received() const override; + bool Send(const DataBuffer& buffer) override; + + // Close immediately, ignoring any queued data or closing procedure. + // This is called when the underlying SctpTransport is being destroyed. + // It is also called by the PeerConnection if SCTP ID assignment fails. + void CloseAbruptlyWithError(RTCError error); + // Specializations of CloseAbruptlyWithError + void CloseAbruptlyWithDataChannelFailure(const std::string& message); + + // Slots for controller to connect signals to. + // + // TODO(deadbeef): Make these private once we're hooking up signals ourselves, + // instead of relying on SctpDataChannelControllerInterface. + + // Called when the SctpTransport's ready to use. That can happen when we've + // finished negotiation, or if the channel was created after negotiation has + // already finished. + void OnTransportReady(bool writable); + + void OnDataReceived(const cricket::ReceiveDataParams& params, + const rtc::CopyOnWriteBuffer& payload); + + // Sets the SCTP sid and adds to transport layer if not set yet. Should only + // be called once. + void SetSctpSid(int sid); + // The remote side started the closing procedure by resetting its outgoing + // stream (our incoming stream). Sets state to kClosing. + void OnClosingProcedureStartedRemotely(int sid); + // The closing procedure is complete; both incoming and outgoing stream + // resets are done and the channel can transition to kClosed. Called + // asynchronously after RemoveSctpDataStream. + void OnClosingProcedureComplete(int sid); + // Called when the transport channel is created. + // Only needs to be called for SCTP data channels. + void OnTransportChannelCreated(); + // Called when the transport channel is unusable. + // This method makes sure the DataChannel is disconnected and changes state + // to kClosed. + void OnTransportChannelClosed(RTCError error); + + DataChannelStats GetStats() const; + + // Emitted when state transitions to kOpen. + sigslot::signal1<DataChannelInterface*> SignalOpened; + // Emitted when state transitions to kClosed. + // This signal can be used to tell when the channel's sid is free. + sigslot::signal1<DataChannelInterface*> SignalClosed; + + // Reset the allocator for internal ID values for testing, so that + // the internal IDs generated are predictable. Test only. + static void ResetInternalIdAllocatorForTesting(int new_value); + + protected: + SctpDataChannel(const InternalDataChannelInit& config, + SctpDataChannelControllerInterface* client, + const std::string& label, + rtc::Thread* signaling_thread, + rtc::Thread* network_thread); + ~SctpDataChannel() override; + + private: + // The OPEN(_ACK) signaling state. + enum HandshakeState { + kHandshakeInit, + kHandshakeShouldSendOpen, + kHandshakeShouldSendAck, + kHandshakeWaitingForAck, + kHandshakeReady + }; + + bool Init(); + void UpdateState(); + void SetState(DataState state); + void DisconnectFromTransport(); + + void DeliverQueuedReceivedData(); + + void SendQueuedDataMessages(); + bool SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked); + bool QueueSendDataMessage(const DataBuffer& buffer); + + void SendQueuedControlMessages(); + void QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer); + bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer); + + rtc::Thread* const signaling_thread_; + rtc::Thread* const network_thread_; + const int internal_id_; + const std::string label_; + const InternalDataChannelInit config_; + DataChannelObserver* observer_ RTC_GUARDED_BY(signaling_thread_) = nullptr; + DataState state_ RTC_GUARDED_BY(signaling_thread_) = kConnecting; + RTCError error_ RTC_GUARDED_BY(signaling_thread_); + uint32_t messages_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; + uint64_t bytes_sent_ RTC_GUARDED_BY(signaling_thread_) = 0; + uint32_t messages_received_ RTC_GUARDED_BY(signaling_thread_) = 0; + uint64_t bytes_received_ RTC_GUARDED_BY(signaling_thread_) = 0; + SctpDataChannelControllerInterface* const controller_ + RTC_GUARDED_BY(signaling_thread_); + bool controller_detached_ RTC_GUARDED_BY(signaling_thread_) = false; + HandshakeState handshake_state_ RTC_GUARDED_BY(signaling_thread_) = + kHandshakeInit; + bool connected_to_transport_ RTC_GUARDED_BY(signaling_thread_) = false; + bool writable_ RTC_GUARDED_BY(signaling_thread_) = false; + // Did we already start the graceful SCTP closing procedure? + bool started_closing_procedure_ RTC_GUARDED_BY(signaling_thread_) = false; + // Control messages that always have to get sent out before any queued + // data. + PacketQueue queued_control_data_ RTC_GUARDED_BY(signaling_thread_); + PacketQueue queued_received_data_ RTC_GUARDED_BY(signaling_thread_); + PacketQueue queued_send_data_ RTC_GUARDED_BY(signaling_thread_); +}; + +// Downcast a PeerConnectionInterface that points to a proxy object +// to its underlying SctpDataChannel object. For testing only. +SctpDataChannel* DowncastProxiedDataChannelInterfaceToSctpDataChannelForTesting( + DataChannelInterface* channel); + +} // namespace webrtc + +#endif // PC_SCTP_DATA_CHANNEL_H_ |