Index: webrtc/base/asyncpacketsocket.h |
diff --git a/webrtc/base/asyncpacketsocket.h b/webrtc/base/asyncpacketsocket.h |
index a5409479511eda1edc8b667b264a891c801380fb..809f1789afedf39e7034b51409de342b4c03438c 100644 |
--- a/webrtc/base/asyncpacketsocket.h |
+++ b/webrtc/base/asyncpacketsocket.h |
@@ -11,133 +11,9 @@ |
#ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |
#define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |
-#include "webrtc/base/constructormagic.h" |
-#include "webrtc/base/dscp.h" |
-#include "webrtc/base/sigslot.h" |
-#include "webrtc/base/socket.h" |
-#include "webrtc/base/timeutils.h" |
-namespace rtc { |
- |
-// This structure holds the info needed to update the packet send time header |
-// extension, including the information needed to update the authentication tag |
-// after changing the value. |
-struct PacketTimeUpdateParams { |
- PacketTimeUpdateParams(); |
- ~PacketTimeUpdateParams(); |
- |
- int rtp_sendtime_extension_id; // extension header id present in packet. |
- std::vector<char> srtp_auth_key; // Authentication key. |
- int srtp_auth_tag_len; // Authentication tag length. |
- int64_t srtp_packet_index; // Required for Rtp Packet authentication. |
-}; |
- |
-// This structure holds meta information for the packet which is about to send |
-// over network. |
-struct PacketOptions { |
- PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {} |
- explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {} |
- |
- DiffServCodePoint dscp; |
- int packet_id; // 16 bits, -1 represents "not set". |
- PacketTimeUpdateParams packet_time_params; |
-}; |
- |
-// This structure will have the information about when packet is actually |
-// received by socket. |
-struct PacketTime { |
- PacketTime() : timestamp(-1), not_before(-1) {} |
- PacketTime(int64_t timestamp, int64_t not_before) |
- : timestamp(timestamp), not_before(not_before) {} |
- |
- int64_t timestamp; // Receive time after socket delivers the data. |
- |
- // Earliest possible time the data could have arrived, indicating the |
- // potential error in the |timestamp| value, in case the system, is busy. For |
- // example, the time of the last select() call. |
- // If unknown, this value will be set to zero. |
- int64_t not_before; |
-}; |
- |
-inline PacketTime CreatePacketTime(int64_t not_before) { |
- return PacketTime(TimeMicros(), not_before); |
-} |
- |
-// Provides the ability to receive packets asynchronously. Sends are not |
-// buffered since it is acceptable to drop packets under high load. |
-class AsyncPacketSocket : public sigslot::has_slots<> { |
- public: |
- enum State { |
- STATE_CLOSED, |
- STATE_BINDING, |
- STATE_BOUND, |
- STATE_CONNECTING, |
- STATE_CONNECTED |
- }; |
- |
- AsyncPacketSocket(); |
- ~AsyncPacketSocket() override; |
- |
- // Returns current local address. Address may be set to null if the |
- // socket is not bound yet (GetState() returns STATE_BINDING). |
- virtual SocketAddress GetLocalAddress() const = 0; |
- |
- // Returns remote address. Returns zeroes if this is not a client TCP socket. |
- virtual SocketAddress GetRemoteAddress() const = 0; |
- |
- // Send a packet. |
- virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0; |
- virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr, |
- const PacketOptions& options) = 0; |
- |
- // Close the socket. |
- virtual int Close() = 0; |
- |
- // Returns current state of the socket. |
- virtual State GetState() const = 0; |
- |
- // Get/set options. |
- virtual int GetOption(Socket::Option opt, int* value) = 0; |
- virtual int SetOption(Socket::Option opt, int value) = 0; |
- |
- // Get/Set current error. |
- // TODO: Remove SetError(). |
- virtual int GetError() const = 0; |
- virtual void SetError(int error) = 0; |
- |
- // Emitted each time a packet is read. Used only for UDP and |
- // connected TCP sockets. |
- sigslot::signal5<AsyncPacketSocket*, const char*, size_t, |
- const SocketAddress&, |
- const PacketTime&> SignalReadPacket; |
- |
- // Emitted each time a packet is sent. |
- sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket; |
- |
- // Emitted when the socket is currently able to send. |
- sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend; |
- |
- // Emitted after address for the socket is allocated, i.e. binding |
- // is finished. State of the socket is changed from BINDING to BOUND |
- // (for UDP and server TCP sockets) or CONNECTING (for client TCP |
- // sockets). |
- sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady; |
- |
- // Emitted for client TCP sockets when state is changed from |
- // CONNECTING to CONNECTED. |
- sigslot::signal1<AsyncPacketSocket*> SignalConnect; |
- |
- // Emitted for client TCP sockets when state is changed from |
- // CONNECTED to CLOSED. |
- sigslot::signal2<AsyncPacketSocket*, int> SignalClose; |
- |
- // Used only for listening TCP sockets. |
- sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection; |
- |
- private: |
- RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket); |
-}; |
- |
-} // namespace rtc |
+// This header is deprecated and is just left here temporarily during |
+// refactoring. See https://bugs.webrtc.org/7634 for more details. |
+#include "webrtc/rtc_base/asyncpacketsocket.h" |
#endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |