Move webrtc/{base => rtc_base}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
diff --git a/webrtc/base/asyncpacketsocket.h b/webrtc/base/asyncpacketsocket.h
index a540947..809f178 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_