/*
 *  Copyright 2004 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 RTC_BASE_VIRTUAL_SOCKET_SERVER_H_
#define RTC_BASE_VIRTUAL_SOCKET_SERVER_H_

#include <deque>
#include <map>
#include <optional>
#include <vector>

#include "api/make_ref_counted.h"
#include "api/ref_counted_base.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/socket_address_pair.h"
#include "rtc_base/socket_server.h"
#include "rtc_base/synchronization/mutex.h"

namespace rtc {

class VirtualSocketPacket;
class VirtualSocketServer;

// Implements the socket interface using the virtual network. Packets are
// passed in tasks using the thread of the socket server.
class VirtualSocket : public Socket, public sigslot::has_slots<> {
 public:
  VirtualSocket(VirtualSocketServer* server, int family, int type);
  ~VirtualSocket() override;

  webrtc::SocketAddress GetLocalAddress() const override;
  webrtc::SocketAddress GetRemoteAddress() const override;

  int Bind(const webrtc::SocketAddress& addr) override;
  int Connect(const webrtc::SocketAddress& addr) override;
  int Close() override;
  int Send(const void* pv, size_t cb) override;
  int SendTo(const void* pv,
             size_t cb,
             const webrtc::SocketAddress& addr) override;
  int Recv(void* pv, size_t cb, int64_t* timestamp) override;
  int RecvFrom(void* pv,
               size_t cb,
               webrtc::SocketAddress* paddr,
               int64_t* timestamp) override;
  int Listen(int backlog) override;
  VirtualSocket* Accept(webrtc::SocketAddress* paddr) override;

  int GetError() const override;
  void SetError(int error) override;
  ConnState GetState() const override;
  int GetOption(Option opt, int* value) override;
  int SetOption(Option opt, int value) override;

  size_t recv_buffer_size() const { return recv_buffer_size_; }
  size_t send_buffer_size() const { return send_buffer_.size(); }
  const char* send_buffer_data() const { return send_buffer_.data(); }

  // Used by server sockets to set the local address without binding.
  void SetLocalAddress(const webrtc::SocketAddress& addr);

  bool was_any() { return was_any_; }
  void set_was_any(bool was_any) { was_any_ = was_any; }

  void SetToBlocked();

  void UpdateRecv(size_t data_size);
  void UpdateSend(size_t data_size);

  void MaybeSignalWriteEvent(size_t capacity);

  // Adds a packet to be sent. Returns delay, based on network_size_.
  uint32_t AddPacket(int64_t cur_time, size_t packet_size);

  int64_t UpdateOrderedDelivery(int64_t ts);

  // Removes stale packets from the network. Returns current size.
  size_t PurgeNetworkPackets(int64_t cur_time);

  void PostPacket(webrtc::TimeDelta delay,
                  std::unique_ptr<VirtualSocketPacket> packet);
  void PostConnect(webrtc::TimeDelta delay,
                   const webrtc::SocketAddress& remote_addr);
  void PostDisconnect(webrtc::TimeDelta delay);

 private:
  // Struct shared with pending tasks that may outlive VirtualSocket.
  class SafetyBlock : public RefCountedNonVirtual<SafetyBlock> {
   public:
    explicit SafetyBlock(VirtualSocket* socket);
    SafetyBlock(const SafetyBlock&) = delete;
    SafetyBlock& operator=(const SafetyBlock&) = delete;
    ~SafetyBlock();

    // Prohibits posted delayed task to access owning VirtualSocket and
    // cleanups members protected by the `mutex`.
    void SetNotAlive();
    bool IsAlive();

    // Copies up to `size` bytes into buffer from the next received packet
    // and fills `addr` with remote address of that received packet.
    // Returns number of bytes copied or negative value on failure.
    int RecvFrom(void* buffer, size_t size, webrtc::SocketAddress& addr);

    void Listen();

    struct AcceptResult {
      int error = 0;
      std::unique_ptr<VirtualSocket> socket;
      webrtc::SocketAddress remote_addr;
    };
    AcceptResult Accept();

    bool AddPacket(std::unique_ptr<VirtualSocketPacket> packet);
    void PostConnect(webrtc::TimeDelta delay,
                     const webrtc::SocketAddress& remote_addr);

   private:
    enum class Signal { kNone, kReadEvent, kConnectEvent };
    // `PostConnect` rely on the fact that std::list iterators are not
    // invalidated on any changes to other elements in the container.
    using PostedConnects = std::list<webrtc::SocketAddress>;

    void PostSignalReadEvent() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
    void MaybeSignalReadEvent();
    Signal Connect(PostedConnects::iterator remote_addr_it);

    webrtc::Mutex mutex_;
    VirtualSocket& socket_;
    bool alive_ RTC_GUARDED_BY(mutex_) = true;
    // Flag indicating if async Task to signal SignalReadEvent is posted.
    // To avoid posting multiple such tasks.
    bool pending_read_signal_event_ RTC_GUARDED_BY(mutex_) = false;

    // Members below do not need to outlive VirtualSocket, but are used by the
    // posted tasks. Keeping them in the VirtualSocket confuses thread
    // annotations because they can't detect that locked mutex is the same mutex
    // this members are guarded by.

    // Addresses of the sockets for potential connect. For each address there
    // is a posted task that should finilze the connect.
    PostedConnects posted_connects_ RTC_GUARDED_BY(mutex_);

    // Data which has been received from the network
    std::list<std::unique_ptr<VirtualSocketPacket>> recv_buffer_
        RTC_GUARDED_BY(mutex_);

    // Pending sockets which can be Accepted
    std::optional<std::deque<webrtc::SocketAddress>> listen_queue_
        RTC_GUARDED_BY(mutex_);
  };

  struct NetworkEntry {
    size_t size;
    int64_t done_time;
  };

  typedef std::deque<NetworkEntry> NetworkQueue;
  typedef std::vector<char> SendBuffer;
  typedef std::map<Option, int> OptionsMap;

  int InitiateConnect(const webrtc::SocketAddress& addr, bool use_delay);
  void CompleteConnect(const webrtc::SocketAddress& addr);
  int SendUdp(const void* pv, size_t cb, const webrtc::SocketAddress& addr);
  int SendTcp(const void* pv, size_t cb);

  void OnSocketServerReadyToSend();

  VirtualSocketServer* const server_;
  const int type_;
  ConnState state_;
  int error_;
  webrtc::SocketAddress local_addr_;
  webrtc::SocketAddress remote_addr_;

  const scoped_refptr<SafetyBlock> safety_ =
      make_ref_counted<SafetyBlock>(this);

  // Data which tcp has buffered for sending
  SendBuffer send_buffer_;
  // Set to false if the last attempt to send resulted in EWOULDBLOCK.
  // Set back to true when the socket can send again.
  bool ready_to_send_ = true;

  // Network model that enforces bandwidth and capacity constraints
  NetworkQueue network_;
  size_t network_size_;
  // The scheduled delivery time of the last packet sent on this socket.
  // It is used to ensure ordered delivery of packets sent on this socket.
  int64_t last_delivery_time_ = 0;

  // The amount of data which is in flight or in recv_buffer_
  size_t recv_buffer_size_;

  // Is this socket bound?
  bool bound_;

  // When we bind a socket to Any, VSS's Bind gives it another address. For
  // dual-stack sockets, we want to distinguish between sockets that were
  // explicitly given a particular address and sockets that had one picked
  // for them by VSS.
  bool was_any_;

  // Store the options that are set
  OptionsMap options_map_;
};

// Simulates a network in the same manner as a loopback interface.  The
// interface can create as many addresses as you want.  All of the sockets
// created by this network will be able to communicate with one another, unless
// they are bound to addresses from incompatible families.
class VirtualSocketServer : public SocketServer {
 public:
  VirtualSocketServer();
  // This constructor needs to be used if the test uses a fake clock and
  // ProcessMessagesUntilIdle, since ProcessMessagesUntilIdle needs a way of
  // advancing time.
  explicit VirtualSocketServer(webrtc::ThreadProcessingFakeClock* fake_clock);
  ~VirtualSocketServer() override;

  VirtualSocketServer(const VirtualSocketServer&) = delete;
  VirtualSocketServer& operator=(const VirtualSocketServer&) = delete;

  // The default source address specifies which local address to use when a
  // socket is bound to the 'any' address, e.g. 0.0.0.0. (If not set, the 'any'
  // address is used as the source address on outgoing virtual packets, exposed
  // to recipient's RecvFrom).
  webrtc::IPAddress GetDefaultSourceAddress(int family);
  void SetDefaultSourceAddress(const webrtc::IPAddress& from_addr);

  // Limits the network bandwidth (maximum bytes per second).  Zero means that
  // all sends occur instantly.  Defaults to 0.
  void set_bandwidth(uint32_t bandwidth) RTC_LOCKS_EXCLUDED(mutex_);

  // Limits the amount of data which can be in flight on the network without
  // packet loss (on a per sender basis).  Defaults to 64 KB.
  void set_network_capacity(uint32_t capacity) RTC_LOCKS_EXCLUDED(mutex_);

  // The amount of data which can be buffered by tcp on the sender's side
  uint32_t send_buffer_capacity() const RTC_LOCKS_EXCLUDED(mutex_);
  void set_send_buffer_capacity(uint32_t capacity) RTC_LOCKS_EXCLUDED(mutex_);

  // The amount of data which can be buffered by tcp on the receiver's side
  uint32_t recv_buffer_capacity() const RTC_LOCKS_EXCLUDED(mutex_);
  void set_recv_buffer_capacity(uint32_t capacity) RTC_LOCKS_EXCLUDED(mutex_);

  // Controls the (transit) delay for packets sent in the network.  This does
  // not inclue the time required to sit in the send queue.  Both of these
  // values are measured in milliseconds.  Defaults to no delay.
  void set_delay_mean(uint32_t delay_mean) RTC_LOCKS_EXCLUDED(mutex_);
  void set_delay_stddev(uint32_t delay_stddev) RTC_LOCKS_EXCLUDED(mutex_);
  void set_delay_samples(uint32_t delay_samples) RTC_LOCKS_EXCLUDED(mutex_);

  // If the (transit) delay parameters are modified, this method should be
  // called to recompute the new distribution.
  void UpdateDelayDistribution() RTC_LOCKS_EXCLUDED(mutex_);

  // Controls the (uniform) probability that any sent packet is dropped.  This
  // is separate from calculations to drop based on queue size.
  void set_drop_probability(double drop_prob) RTC_LOCKS_EXCLUDED(mutex_);

  // Controls the maximum UDP payload for the networks simulated
  // by this server. Any UDP payload sent that is larger than this will
  // be dropped.
  void set_max_udp_payload(size_t payload_size) RTC_LOCKS_EXCLUDED(mutex_);

  // If `blocked` is true, subsequent attempts to send will result in -1 being
  // returned, with the socket error set to EWOULDBLOCK.
  //
  // If this method is later called with `blocked` set to false, any sockets
  // that previously failed to send with EWOULDBLOCK will emit SignalWriteEvent.
  //
  // This can be used to simulate the send buffer on a network interface being
  // full, and test functionality related to EWOULDBLOCK/SignalWriteEvent.
  void SetSendingBlocked(bool blocked) RTC_LOCKS_EXCLUDED(mutex_);

  // SocketFactory:
  VirtualSocket* CreateSocket(int family, int type) override;

  // SocketServer:
  void SetMessageQueue(Thread* queue) override;
  bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override;
  void WakeUp() override;

  void SetDelayOnAddress(const webrtc::SocketAddress& address, int delay_ms) {
    delay_by_ip_[address.ipaddr()] = delay_ms;
  }

  // Used by TurnPortTest and TcpPortTest (for example), to mimic a case where
  // a proxy returns the local host address instead of the original one the
  // port was bound against. Please see WebRTC issue 3927 for more detail.
  //
  // If SetAlternativeLocalAddress(A, B) is called, then when something
  // attempts to bind a socket to address A, it will get a socket bound to
  // address B instead.
  void SetAlternativeLocalAddress(const webrtc::IPAddress& address,
                                  const webrtc::IPAddress& alternative);

  typedef std::pair<double, double> Point;
  typedef std::vector<Point> Function;

  static std::unique_ptr<Function> CreateDistribution(uint32_t mean,
                                                      uint32_t stddev,
                                                      uint32_t samples);

  // Similar to Thread::ProcessMessages, but it only processes messages until
  // there are no immediate messages or pending network traffic.  Returns false
  // if Thread::Stop() was called.
  bool ProcessMessagesUntilIdle();

  // Sets the next port number to use for testing.
  void SetNextPortForTesting(uint16_t port);

  // Close a pair of Tcp connections by addresses. Both connections will have
  // its own OnClose invoked.
  bool CloseTcpConnections(const webrtc::SocketAddress& addr_local,
                           const webrtc::SocketAddress& addr_remote);

  // Number of packets that clients have attempted to send through this virtual
  // socket server. Intended to be used for test assertions.
  uint32_t sent_packets() const RTC_LOCKS_EXCLUDED(mutex_);

  // Assign IP and Port if application's address is unspecified. Also apply
  // `alternative_address_mapping_`.
  webrtc::SocketAddress AssignBindAddress(
      const webrtc::SocketAddress& app_addr);

  // Binds the given socket to the given (fully-defined) address.
  int Bind(VirtualSocket* socket, const webrtc::SocketAddress& addr);

  int Unbind(const webrtc::SocketAddress& addr, VirtualSocket* socket);

  // Adds a mapping between this socket pair and the socket.
  void AddConnection(const webrtc::SocketAddress& client,
                     const webrtc::SocketAddress& server,
                     VirtualSocket* socket);

  // Connects the given socket to the socket at the given address
  int Connect(VirtualSocket* socket,
              const webrtc::SocketAddress& remote_addr,
              bool use_delay);

  // Sends a disconnect message to the socket at the given address
  bool Disconnect(VirtualSocket* socket);

  // Lookup address, and disconnect corresponding socket.
  bool Disconnect(const webrtc::SocketAddress& addr);

  // Lookup connection, close corresponding socket.
  bool Disconnect(const webrtc::SocketAddress& local_addr,
                  const webrtc::SocketAddress& remote_addr);

  // Sends the given packet to the socket at the given address (if one exists).
  int SendUdp(VirtualSocket* socket,
              const char* data,
              size_t data_size,
              const webrtc::SocketAddress& remote_addr);

  // Moves as much data as possible from the sender's buffer to the network
  void SendTcp(VirtualSocket* socket) RTC_LOCKS_EXCLUDED(mutex_);

  // Like above, but lookup sender by address.
  void SendTcp(const webrtc::SocketAddress& addr) RTC_LOCKS_EXCLUDED(mutex_);

  // Computes the number of milliseconds required to send a packet of this size.
  uint32_t SendDelay(uint32_t size) RTC_LOCKS_EXCLUDED(mutex_);

  // Sending was previously blocked, but now isn't.
  sigslot::signal0<> SignalReadyToSend;

 protected:
  // Returns a new IP not used before in this network.
  webrtc::IPAddress GetNextIP(int family);

  // Find the socket bound to the given address
  VirtualSocket* LookupBinding(const webrtc::SocketAddress& addr);

 private:
  friend VirtualSocket;
  uint16_t GetNextPort();

  // Find the socket pair corresponding to this server address.
  VirtualSocket* LookupConnection(const webrtc::SocketAddress& client,
                                  const webrtc::SocketAddress& server);

  void RemoveConnection(const webrtc::SocketAddress& client,
                        const webrtc::SocketAddress& server);

  // Places a packet on the network.
  void AddPacketToNetwork(VirtualSocket* socket,
                          VirtualSocket* recipient,
                          int64_t cur_time,
                          const char* data,
                          size_t data_size,
                          size_t header_size,
                          bool ordered);

  // If the delay has been set for the address of the socket, returns the set
  // delay. Otherwise, returns a random transit delay chosen from the
  // appropriate distribution.
  uint32_t GetTransitDelay(Socket* socket);

  // Basic operations on functions.
  static std::unique_ptr<Function> Accumulate(std::unique_ptr<Function> f);
  static std::unique_ptr<Function> Invert(std::unique_ptr<Function> f);
  static std::unique_ptr<Function> Resample(std::unique_ptr<Function> f,
                                            double x1,
                                            double x2,
                                            uint32_t samples);
  static double Evaluate(const Function* f, double x);

  // Determine if two sockets should be able to communicate.
  // We don't (currently) specify an address family for sockets; instead,
  // the currently bound address is used to infer the address family.
  // Any socket that is not explicitly bound to an IPv4 address is assumed to be
  // dual-stack capable.
  // This function tests if two addresses can communicate, as well as the
  // sockets to which they may be bound (the addresses may or may not yet be
  // bound to the sockets).
  // First the addresses are tested (after normalization):
  // If both have the same family, then communication is OK.
  // If only one is IPv4 then false, unless the other is bound to ::.
  // This applies even if the IPv4 address is 0.0.0.0.
  // The socket arguments are optional; the sockets are checked to see if they
  // were explicitly bound to IPv6-any ('::'), and if so communication is
  // permitted.
  // NB: This scheme doesn't permit non-dualstack IPv6 sockets.
  static bool CanInteractWith(VirtualSocket* local, VirtualSocket* remote);

  typedef std::map<webrtc::SocketAddress, VirtualSocket*> AddressMap;
  typedef std::map<webrtc::SocketAddressPair, VirtualSocket*> ConnectionMap;

  // May be null if the test doesn't use a fake clock, or it does but doesn't
  // use ProcessMessagesUntilIdle.
  webrtc::ThreadProcessingFakeClock* fake_clock_ = nullptr;

  // Used to implement Wait/WakeUp.
  Event wakeup_;
  Thread* msg_queue_;
  bool stop_on_idle_;
  in_addr next_ipv4_;
  in6_addr next_ipv6_;
  uint16_t next_port_;
  AddressMap* bindings_;
  ConnectionMap* connections_;

  webrtc::IPAddress default_source_address_v4_;
  webrtc::IPAddress default_source_address_v6_;

  mutable webrtc::Mutex mutex_;

  uint32_t bandwidth_ RTC_GUARDED_BY(mutex_);
  uint32_t network_capacity_ RTC_GUARDED_BY(mutex_);
  uint32_t send_buffer_capacity_ RTC_GUARDED_BY(mutex_);
  uint32_t recv_buffer_capacity_ RTC_GUARDED_BY(mutex_);
  uint32_t delay_mean_ RTC_GUARDED_BY(mutex_);
  uint32_t delay_stddev_ RTC_GUARDED_BY(mutex_);
  uint32_t delay_samples_ RTC_GUARDED_BY(mutex_);

  // Used for testing.
  uint32_t sent_packets_ RTC_GUARDED_BY(mutex_) = 0;

  std::map<webrtc::IPAddress, int> delay_by_ip_;
  std::map<webrtc::IPAddress, webrtc::IPAddress> alternative_address_mapping_;
  std::unique_ptr<Function> delay_dist_;

  double drop_prob_ RTC_GUARDED_BY(mutex_);
  // The largest UDP payload permitted on this virtual socket server.
  // The default is the max size of IPv4 fragmented UDP packet payload:
  // 65535 bytes - 8 bytes UDP header - 20 bytes IP header.
  size_t max_udp_payload_ RTC_GUARDED_BY(mutex_) = 65507;

  bool sending_blocked_ RTC_GUARDED_BY(mutex_) = false;
};

}  // namespace rtc

#endif  // RTC_BASE_VIRTUAL_SOCKET_SERVER_H_