From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 7 Apr 2024 21:33:14 +0200
Subject: Adding upstream version 115.7.0esr.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../libwebrtc/p2p/base/packet_transport_internal.h | 108 +++++++++++++++++++++
 1 file changed, 108 insertions(+)
 create mode 100644 third_party/libwebrtc/p2p/base/packet_transport_internal.h

(limited to 'third_party/libwebrtc/p2p/base/packet_transport_internal.h')

diff --git a/third_party/libwebrtc/p2p/base/packet_transport_internal.h b/third_party/libwebrtc/p2p/base/packet_transport_internal.h
new file mode 100644
index 0000000000..2ca47d533d
--- /dev/null
+++ b/third_party/libwebrtc/p2p/base/packet_transport_internal.h
@@ -0,0 +1,108 @@
+/*
+ *  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 <vector>
+
+#include "absl/types/optional.h"
+#include "p2p/base/port.h"
+#include "rtc_base/async_packet_socket.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;
+
+  // Signalled each time a packet is received on this channel.
+  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;
+};
+
+}  // namespace rtc
+
+#endif  // P2P_BASE_PACKET_TRANSPORT_INTERNAL_H_
-- 
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