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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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