Revert "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)

Will reland in two different commits to preserve git blame history.

BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org

Change-Id: I550da8525aeb9c5b8f96338fcf1c9714f3dcdab1
Reviewed-on: https://chromium-review.googlesource.com/554610
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18820}

webrtc/base/asyncpacketsocket.h

@@ -11,9 +11,133 @@
 #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"
 
-// 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"
+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
 
 #endif  // WEBRTC_BASE_ASYNCPACKETSOCKET_H_