| /* |
| * 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 <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "p2p/base/port.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/callback_list.h" |
| #include "rtc_base/network/received_packet.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; |
| |
| template <typename F> |
| void RegisterReceivedPacketCallback(void* id, F&& callback) { |
| RTC_DCHECK_RUN_ON(&network_checker_); |
| received_packet_callback_list_.AddReceiver(id, std::forward<F>(callback)); |
| } |
| void DeregisterReceivedPacketCallback(void* id) { |
| RTC_DCHECK_RUN_ON(&network_checker_); |
| received_packet_callback_list_.RemoveReceivers(id); |
| } |
| |
| // Signalled each time a packet is received on this channel. |
| // TODO(bugs.webrtc.org:15368): Deprecate and remove. Replace with |
| // RegisterReceivedPacketCallback. |
| 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; |
| |
| void NotifyPacketReceived(const rtc::ReceivedPacket& packet); |
| |
| webrtc::SequenceChecker network_checker_{webrtc::SequenceChecker::kDetached}; |
| |
| private: |
| webrtc::CallbackList<PacketTransportInternal*, const rtc::ReceivedPacket&> |
| received_packet_callback_list_ RTC_GUARDED_BY(&network_checker_); |
| }; |
| |
| } // namespace rtc |
| |
| #endif // P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_ |