109 lines
4.1 KiB
C++
109 lines
4.1 KiB
C++
/*
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* Copyright 2017 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
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#define P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "p2p/base/port.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/socket.h"
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#include "rtc_base/system/rtc_export.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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namespace rtc {
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struct PacketOptions;
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struct SentPacket;
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class RTC_EXPORT PacketTransportInternal : public sigslot::has_slots<> {
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public:
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virtual const std::string& transport_name() const = 0;
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// The transport has been established.
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virtual bool writable() const = 0;
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// The transport has received a packet in the last X milliseconds, here X is
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// configured by each implementation.
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virtual bool receiving() const = 0;
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// Attempts to send the given packet.
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// The return value is < 0 on failure. The return value in failure case is not
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// descriptive. Depending on failure cause and implementation details
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// GetError() returns an descriptive errno.h error value.
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// This mimics posix socket send() or sendto() behavior.
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// TODO(johan): Reliable, meaningful, consistent error codes for all
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// implementations would be nice.
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// TODO(johan): Remove the default argument once channel code is updated.
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virtual int SendPacket(const char* data,
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size_t len,
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const rtc::PacketOptions& options,
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int flags = 0) = 0;
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// Sets a socket option. Note that not all options are
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// supported by all transport types.
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virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
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// TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements
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// this, remove the default implementation.
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virtual bool GetOption(rtc::Socket::Option opt, int* value);
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// Returns the most recent error that occurred on this channel.
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virtual int GetError() = 0;
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// Returns the current network route with transport overhead.
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// TODO(zhihuang): Make it pure virtual once the Chrome/remoting is updated.
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virtual absl::optional<NetworkRoute> network_route() const;
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// Emitted when the writable state, represented by |writable()|, changes.
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sigslot::signal1<PacketTransportInternal*> SignalWritableState;
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// Emitted when the PacketTransportInternal is ready to send packets. "Ready
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// to send" is more sensitive than the writable state; a transport may be
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// writable, but temporarily not able to send packets. For example, the
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// underlying transport's socket buffer may be full, as indicated by
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// SendPacket's return code and/or GetError.
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sigslot::signal1<PacketTransportInternal*> SignalReadyToSend;
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// Emitted when receiving state changes to true.
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sigslot::signal1<PacketTransportInternal*> SignalReceivingState;
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// Signalled each time a packet is received on this channel.
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sigslot::signal5<PacketTransportInternal*,
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const char*,
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size_t,
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// TODO(bugs.webrtc.org/9584): Change to passing the int64_t
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// timestamp by value.
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const int64_t&,
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int>
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SignalReadPacket;
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// Signalled each time a packet is sent on this channel.
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sigslot::signal2<PacketTransportInternal*, const rtc::SentPacket&>
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SignalSentPacket;
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// Signalled when the current network route has changed.
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sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
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// Signalled when the transport is closed.
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sigslot::signal1<PacketTransportInternal*> SignalClosed;
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protected:
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PacketTransportInternal();
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~PacketTransportInternal() override;
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};
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} // namespace rtc
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#endif // P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
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