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