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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/p2p/base/tcpport.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
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+/*
+ * 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 WEBRTC_P2P_BASE_TCPPORT_H_
+#define WEBRTC_P2P_BASE_TCPPORT_H_
+
+#include <list>
+#include <string>
+#include "webrtc/p2p/base/port.h"
+#include "webrtc/base/asyncpacketsocket.h"
+
+namespace cricket {
+
+class TCPConnection;
+
+// Communicates using a local TCP port.
+//
+// This class is designed to allow subclasses to take advantage of the
+// connection management provided by this class. A subclass should take of all
+// packet sending and preparation, but when a packet is received, it should
+// call this TCPPort::OnReadPacket (3 arg) to dispatch to a connection.
+class TCPPort : public Port {
+ public:
+ static TCPPort* Create(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16_t min_port,
+ uint16_t max_port,
+ const std::string& username,
+ const std::string& password,
+ bool allow_listen) {
+ TCPPort* port = new TCPPort(thread, factory, network, ip, min_port,
+ max_port, username, password, allow_listen);
+ if (!port->Init()) {
+ delete port;
+ port = NULL;
+ }
+ return port;
+ }
+ virtual ~TCPPort();
+
+ virtual Connection* CreateConnection(const Candidate& address,
+ CandidateOrigin origin);
+
+ virtual void PrepareAddress();
+
+ virtual int GetOption(rtc::Socket::Option opt, int* value);
+ virtual int SetOption(rtc::Socket::Option opt, int value);
+ virtual int GetError();
+
+ protected:
+ TCPPort(rtc::Thread* thread,
+ rtc::PacketSocketFactory* factory,
+ rtc::Network* network,
+ const rtc::IPAddress& ip,
+ uint16_t min_port,
+ uint16_t max_port,
+ const std::string& username,
+ const std::string& password,
+ bool allow_listen);
+ bool Init();
+
+ // Handles sending using the local TCP socket.
+ virtual int SendTo(const void* data, size_t size,
+ const rtc::SocketAddress& addr,
+ const rtc::PacketOptions& options,
+ bool payload);
+
+ // Accepts incoming TCP connection.
+ void OnNewConnection(rtc::AsyncPacketSocket* socket,
+ rtc::AsyncPacketSocket* new_socket);
+
+ private:
+ struct Incoming {
+ rtc::SocketAddress addr;
+ rtc::AsyncPacketSocket* socket;
+ };
+
+ rtc::AsyncPacketSocket* GetIncoming(
+ const rtc::SocketAddress& addr, bool remove = false);
+
+ // Receives packet signal from the local TCP Socket.
+ void OnReadPacket(rtc::AsyncPacketSocket* socket,
+ const char* data, size_t size,
+ const rtc::SocketAddress& remote_addr,
+ const rtc::PacketTime& packet_time);
+
+ void OnReadyToSend(rtc::AsyncPacketSocket* socket);
+
+ void OnAddressReady(rtc::AsyncPacketSocket* socket,
+ const rtc::SocketAddress& address);
+
+ // TODO: Is this still needed?
+ bool incoming_only_;
+ bool allow_listen_;
+ rtc::AsyncPacketSocket* socket_;
+ int error_;
+ std::list<Incoming> incoming_;
+
+ friend class TCPConnection;
+};
+
+class TCPConnection : public Connection {
+ public:
+ // Connection is outgoing unless socket is specified
+ TCPConnection(TCPPort* port, const Candidate& candidate,
+ rtc::AsyncPacketSocket* socket = 0);
+ virtual ~TCPConnection();
+
+ virtual int Send(const void* data, size_t size,
+ const rtc::PacketOptions& options);
+ virtual int GetError();
+
+ rtc::AsyncPacketSocket* socket() { return socket_.get(); }
+
+ void OnMessage(rtc::Message* pmsg);
+
+ // Allow test cases to overwrite the default timeout period.
+ int reconnection_timeout() const { return reconnection_timeout_; }
+ void set_reconnection_timeout(int timeout_in_ms) {
+ reconnection_timeout_ = timeout_in_ms;
+ }
+
+ protected:
+ enum {
+ MSG_TCPCONNECTION_DELAYED_ONCLOSE = Connection::MSG_FIRST_AVAILABLE,
+ };
+
+ // Set waiting_for_stun_binding_complete_ to false to allow data packets in
+ // addition to what Port::OnConnectionRequestResponse does.
+ virtual void OnConnectionRequestResponse(ConnectionRequest* req,
+ StunMessage* response);
+
+ private:
+ // Helper function to handle the case when Ping or Send fails with error
+ // related to socket close.
+ void MaybeReconnect();
+
+ void CreateOutgoingTcpSocket();
+
+ void ConnectSocketSignals(rtc::AsyncPacketSocket* socket);
+
+ void OnConnect(rtc::AsyncPacketSocket* socket);
+ void OnClose(rtc::AsyncPacketSocket* socket, int error);
+ void OnReadPacket(rtc::AsyncPacketSocket* socket,
+ const char* data, size_t size,
+ const rtc::SocketAddress& remote_addr,
+ const rtc::PacketTime& packet_time);
+ void OnReadyToSend(rtc::AsyncPacketSocket* socket);
+
+ rtc::scoped_ptr<rtc::AsyncPacketSocket> socket_;
+ int error_;
+ bool outgoing_;
+
+ // Guard against multiple outgoing tcp connection during a reconnect.
+ bool connection_pending_;
+
+ // Guard against data packets sent when we reconnect a TCP connection. During
+ // reconnecting, when a new tcp connection has being made, we can't send data
+ // packets out until the STUN binding is completed (i.e. the write state is
+ // set to WRITABLE again by Connection::OnConnectionRequestResponse). IPC
+ // socket, when receiving data packets before that, will trigger OnError which
+ // will terminate the newly created connection.
+ bool pretending_to_be_writable_;
+
+ // Allow test case to overwrite the default timeout period.
+ int reconnection_timeout_;
+
+ friend class TCPPort;
+};
+
+} // namespace cricket
+
+#endif // WEBRTC_P2P_BASE_TCPPORT_H_