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/rtc_base/asyncpacketsocket.h b/webrtc/rtc_base/asyncpacketsocket.h
new file mode 100644
index 0000000..5a6c97e
--- /dev/null
+++ b/webrtc/rtc_base/asyncpacketsocket.h
@@ -0,0 +1,143 @@
+/*
+ * Copyright 2004 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 WEBRTC_RTC_BASE_ASYNCPACKETSOCKET_H_
+#define WEBRTC_RTC_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
+
+#endif // WEBRTC_RTC_BASE_ASYNCPACKETSOCKET_H_