Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 305 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..13bebd4612
--- /dev/null
+++ b/third_party/libwebrtc/pc/sctp_data_channel.h
@@ -0,0 +1,305 @@
+/*
+ *  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/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "pc/data_channel_utils.h"
+#include "pc/sctp_utils.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/ssl_stream_adapter.h"  // For SSLRole
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/weak_ptr.h"
+
+namespace webrtc {
+
+class SctpDataChannel;
+
+// Interface that acts as a bridge from the data channel to the transport.
+// All methods in this interface need to be invoked on the network thread.
+class SctpDataChannelControllerInterface {
+ public:
+  // Sends the data to the transport.
+  virtual RTCError SendData(StreamId sid,
+                            const SendDataParams& params,
+                            const rtc::CopyOnWriteBuffer& payload) = 0;
+  // Adds the data channel SID to the transport for SCTP.
+  virtual void AddSctpDataStream(StreamId 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(StreamId sid) = 0;
+  // Notifies the controller of state changes.
+  virtual void OnChannelStateChanged(SctpDataChannel* data_channel,
+                                     DataChannelInterface::DataState state) = 0;
+
+ protected:
+  virtual ~SctpDataChannelControllerInterface() {}
+};
+
+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);
+
+  // Does basic validation to determine if a data channel instance can be
+  // constructed using the configuration.
+  bool IsValid() const;
+
+  OpenHandshakeRole open_handshake_role;
+  // Optional fallback or backup flag from PC that's used for non-prenegotiated
+  // stream ids in situations where we cannot determine the SSL role from the
+  // transport for purposes of generating a stream ID.
+  // See: https://www.rfc-editor.org/rfc/rfc8832.html#name-protocol-overview
+  absl::optional<rtc::SSLRole> fallback_ssl_role;
+};
+
+// Helper class to allocate unique IDs for SCTP DataChannels.
+class SctpSidAllocator {
+ public:
+  SctpSidAllocator() = default;
+  // 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.
+  // If a `StreamId` cannot be allocated, `StreamId::HasValue()` will be false.
+  StreamId AllocateSid(rtc::SSLRole role);
+
+  // Attempts to reserve a specific sid. Returns false if it's unavailable.
+  bool ReserveSid(StreamId sid);
+
+  // Indicates that `sid` isn't in use any more, and is thus available again.
+  void ReleaseSid(StreamId sid);
+
+ private:
+  flat_set<StreamId> used_sids_ RTC_GUARDED_BY(&sequence_checker_);
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_{
+      SequenceChecker::kDetached};
+};
+
+// SctpDataChannel is an implementation of the DataChannelInterface based on
+// SctpTransport. It provides an implementation of unreliable or
+// reliable data 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:
+  static rtc::scoped_refptr<SctpDataChannel> Create(
+      rtc::WeakPtr<SctpDataChannelControllerInterface> controller,
+      const std::string& label,
+      bool connected_to_transport,
+      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.
+  // The `signaling_safety` flag is used for the ObserverAdapter callback proxy
+  // which delivers callbacks on the signaling thread but must not deliver such
+  // callbacks after the peerconnection has been closed. The data controller
+  // will update the flag when closed, which will cancel any pending event
+  // notifications.
+  static rtc::scoped_refptr<DataChannelInterface> CreateProxy(
+      rtc::scoped_refptr<SctpDataChannel> channel,
+      rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_safety);
+
+  void RegisterObserver(DataChannelObserver* observer) override;
+  void UnregisterObserver() override;
+
+  std::string label() const override;
+  bool reliable() const override;
+  bool ordered() const override;
+
+  // Backwards compatible accessors
+  uint16_t maxRetransmitTime() const override;
+  uint16_t maxRetransmits() const override;
+
+  absl::optional<int> maxPacketLifeTime() const override;
+  absl::optional<int> maxRetransmitsOpt() const override;
+  std::string protocol() const override;
+  bool negotiated() const override;
+  int id() const override;
+  Priority priority() const override;
+
+  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;
+  void SendAsync(DataBuffer buffer,
+                 absl::AnyInvocable<void(RTCError) &&> on_complete) 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);
+
+  // 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();
+
+  void OnDataReceived(DataMessageType type,
+                      const rtc::CopyOnWriteBuffer& payload);
+
+  // Sets the SCTP sid and adds to transport layer if not set yet. Should only
+  // be called once.
+  void SetSctpSid_n(StreamId sid);
+
+  // The remote side started the closing procedure by resetting its outgoing
+  // stream (our incoming stream). Sets state to kClosing.
+  void OnClosingProcedureStartedRemotely();
+  // 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();
+  // Called when the transport channel is created.
+  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;
+
+  // Returns a unique identifier that's guaranteed to always be available,
+  // doesn't change throughout SctpDataChannel's lifetime and is used for
+  // stats purposes (see also `GetStats()`).
+  int internal_id() const { return internal_id_; }
+
+  StreamId sid_n() const {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    return id_n_;
+  }
+
+  // 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,
+                  rtc::WeakPtr<SctpDataChannelControllerInterface> controller,
+                  const std::string& label,
+                  bool connected_to_transport,
+                  rtc::Thread* signaling_thread,
+                  rtc::Thread* network_thread);
+  ~SctpDataChannel() override;
+
+ private:
+  class ObserverAdapter;
+
+  // The OPEN(_ACK) signaling state.
+  enum HandshakeState {
+    kHandshakeInit,
+    kHandshakeShouldSendOpen,
+    kHandshakeShouldSendAck,
+    kHandshakeWaitingForAck,
+    kHandshakeReady
+  };
+
+  RTCError SendImpl(DataBuffer buffer) RTC_RUN_ON(network_thread_);
+  void UpdateState() RTC_RUN_ON(network_thread_);
+  void SetState(DataState state) RTC_RUN_ON(network_thread_);
+
+  void DeliverQueuedReceivedData() RTC_RUN_ON(network_thread_);
+
+  void SendQueuedDataMessages() RTC_RUN_ON(network_thread_);
+  RTCError SendDataMessage(const DataBuffer& buffer, bool queue_if_blocked)
+      RTC_RUN_ON(network_thread_);
+  bool QueueSendDataMessage(const DataBuffer& buffer)
+      RTC_RUN_ON(network_thread_);
+
+  void SendQueuedControlMessages() RTC_RUN_ON(network_thread_);
+  bool SendControlMessage(const rtc::CopyOnWriteBuffer& buffer)
+      RTC_RUN_ON(network_thread_);
+
+  bool connected_to_transport() const RTC_RUN_ON(network_thread_) {
+    return network_safety_->alive();
+  }
+
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const network_thread_;
+  StreamId id_n_ RTC_GUARDED_BY(network_thread_);
+  const int internal_id_;
+  const std::string label_;
+  const std::string protocol_;
+  const absl::optional<int> max_retransmit_time_;
+  const absl::optional<int> max_retransmits_;
+  const absl::optional<Priority> priority_;
+  const bool negotiated_;
+  const bool ordered_;
+
+  DataChannelObserver* observer_ RTC_GUARDED_BY(network_thread_) = nullptr;
+  std::unique_ptr<ObserverAdapter> observer_adapter_;
+  DataState state_ RTC_GUARDED_BY(network_thread_) = kConnecting;
+  RTCError error_ RTC_GUARDED_BY(network_thread_);
+  uint32_t messages_sent_ RTC_GUARDED_BY(network_thread_) = 0;
+  uint64_t bytes_sent_ RTC_GUARDED_BY(network_thread_) = 0;
+  uint32_t messages_received_ RTC_GUARDED_BY(network_thread_) = 0;
+  uint64_t bytes_received_ RTC_GUARDED_BY(network_thread_) = 0;
+  rtc::WeakPtr<SctpDataChannelControllerInterface> controller_
+      RTC_GUARDED_BY(network_thread_);
+  HandshakeState handshake_state_ RTC_GUARDED_BY(network_thread_) =
+      kHandshakeInit;
+  // Did we already start the graceful SCTP closing procedure?
+  bool started_closing_procedure_ RTC_GUARDED_BY(network_thread_) = false;
+  // Control messages that always have to get sent out before any queued
+  // data.
+  PacketQueue queued_control_data_ RTC_GUARDED_BY(network_thread_);
+  PacketQueue queued_received_data_ RTC_GUARDED_BY(network_thread_);
+  PacketQueue queued_send_data_ RTC_GUARDED_BY(network_thread_);
+  rtc::scoped_refptr<PendingTaskSafetyFlag> network_safety_ =
+      PendingTaskSafetyFlag::CreateDetachedInactive();
+};
+
+}  // namespace webrtc
+
+#endif  // PC_SCTP_DATA_CHANNEL_H_