Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )

Reason for revert:
Broke Chrome:

https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_chromeos_compile_dbg_ng/builds/143025/steps/compile%20%28with%20patch%29/logs/stdio

FAILED: cd ../../third_party/libjingle; python ../../native_client/build/build_nexe.py --root ../.. --product-dir ../../out/Debug/xyz --config-name Debug -t ../../native_client/toolchain/ --arch pnacl --build newlib_plib --name ../../out/Debug/gen/tc_pnacl_newlib/lib/libjingle_nacl.a --objdir ../../out/Debug/obj/third_party/libjingle/libjingle_nacl.gen/pnacl_newlib-pnacl/libjingle_nacl "--include-dirs=../../out/Debug/gen/tc_pnacl_newlib/include ../.. \"../../out/Debug/gen\" ./source ../ ../../native_client_sdk/src/libraries ../../native_client_sdk/src/libraries/nacl_io/include ../../native_client_sdk/src/libraries/third_party/newlib-extras ../expat/files/lib ../boringssl/src/include" "--compile_flags=-O2 -g -Wall -fdiagnostics-show-option -Werror  -Wno-unused-function -Wno-char-subscripts -Wno-c++11-extensions -Wno-unnamed-type-template-args -Wno-extra-semi -Wno-unused-private-field -Wno-char-subscripts -Wno-unused-function \"-std=gnu++11\" " --gomadir /b/build/goma "--defines=\"__STDC_LIMIT_MACROS=1\" \"__STDC_FORMAT_MACROS=1\" \"_GNU_SOURCE=1\" \"_POSIX_C_SOURCE=199506\" \"_XOPEN_SOURCE=600\" \"DYNAMIC_ANNOTATIONS_ENABLED=1\" \"DYNAMIC_ANNOTATIONS_PREFIX=NACL_\" \"NACL_BUILD_ARCH=x86\" V8_DEPRECATION_WARNINGS \"CLD_VERSION=2\" \"_FILE_OFFSET_BITS=64\" CHROMIUM_BUILD \"CR_CLANG_REVISION=255169-1\" COMPONENT_BUILD UI_COMPOSITOR_IMAGE_TRANSPORT \"USE_AURA=1\" \"USE_ASH=1\" \"USE_PANGO=1\" \"USE_CAIRO=1\" \"USE_DEFAULT_RENDER_THEME=1\" \"USE_LIBJPEG_TURBO=1\" \"USE_X11=1\" \"IMAGE_LOADER_EXTENSION=1\" \"ENABLE_WEBRTC=1\" \"ENABLE_MEDIA_ROUTER=1\" USE_PROPRIETARY_CODECS ENABLE_PEPPER_CDMS ENABLE_CONFIGURATION_POLICY ENABLE_NOTIFICATIONS \"ENABLE_HIDPI=1\" \"ENABLE_TOPCHROME_MD=1\" USE_UDEV DONT_EMBED_BUILD_METADATA \"DCHECK_ALWAYS_ON=1\" FIELDTRIAL_TESTING_ENABLED \"ENABLE_TASK_MANAGER=1\" \"ENABLE_EXTENSIONS=1\" \"ENABLE_PDF=1\" \"ENABLE_PLUGINS=1\" \"ENABLE_SESSION_SERVICE=1\" \"ENABLE_THEMES=1\" \"ENABLE_AUTOFILL_DIALOG=1\" \"ENABLE_BACKGROUND=1\" \"ENABLE_PRINTING=1\" \"ENABLE_PRINT_PREVIEW=1\" \"ENABLE_SPELLCHECK=1\" \"ENABLE_CAPTIVE_PORTAL_DETECTION=1\" \"ENABLE_APP_LIST=1\" \"ENABLE_SUPERVISED_USERS=1\" \"ENABLE_MDNS=1\" \"ENABLE_SERVICE_DISCOVERY=1\" V8_USE_EXTERNAL_STARTUP_DATA FULL_SAFE_BROWSING SAFE_BROWSING_CSD SAFE_BROWSING_DB_LOCAL EXPAT_RELATIVE_PATH FEATURE_ENABLE_SSL GTEST_RELATIVE_PATH HAVE_OPENSSL_SSL_H NO_MAIN_THREAD_WRAPPING NO_SOUND_SYSTEM WEBRTC_POSIX SRTP_RELATIVE_PATH SSL_USE_OPENSSL USE_WEBRTC_DEV_BRANCH \"timezone=_timezone\" XML_STATIC \"USE_LIBPCI=1\" \"USE_OPENSSL=1\" \"USE_OPENSSL_CERTS=1\"" "--link_flags=-B../../out/Debug/gen/tc_pnacl_newlib/lib  " "--source-list=../../out/gypfiles/third_party/libjingle/pnacl_newlib.libjingle_nacl.source_list.gypcmd"
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:50:23: error: 'CreateConnection' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
  virtual Connection* CreateConnection(const Candidate& address,
                      ^
../webrtc/p2p/base/portinterface.h:71:23: note: overridden virtual function is here
  virtual Connection* CreateConnection(
                      ^
In file included from ../webrtc/p2p/base/tcpport.cc:67:
../webrtc/p2p/base/tcpport.h:53:16: error: 'PrepareAddress' overrides a member function but is not marked 'override' [-Werror,-Winconsistent-missing-override]
  virtual void PrepareAddress();
               ^
../webrtc/p2p/base/portinterface.h:63:16: note: overridden virtual function is here
  virtual void PrepareAddress() = 0;
               ^

(etc)

Original issue's description:
> Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
>
> To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
>
> BUG=4173
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/7307952a5bf63311e5f9a2a90089a06177e42716
> Cr-Commit-Position: refs/heads/master@{#11247}

TBR=pthatcher@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=4173

Review URL: https://codereview.webrtc.org/1586063002

Cr-Commit-Position: refs/heads/master@{#11249}

13 files changed, 12 insertions, 39 deletions

diff --git a/webrtc/p2p/base/p2ptransportchannel.cc b/webrtc/p2p/base/p2ptransportchannel.cc
index 952cfab747..c58a235f2d 100644
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@@ -1424,7 +1424,8 @@ void P2PTransportChannel::OnReadPacket(Connection* connection,
   }
 }
 
-void P2PTransportChannel::OnSentPacket(const rtc::SentPacket& sent_packet) {
+void P2PTransportChannel::OnSentPacket(PortInterface* port,
+                                       const rtc::SentPacket& sent_packet) {
   ASSERT(worker_thread_ == rtc::Thread::Current());
 
   SignalSentPacket(this, sent_packet);
diff --git a/webrtc/p2p/base/p2ptransportchannel.h b/webrtc/p2p/base/p2ptransportchannel.h
index f2e9315343..3927b17659 100644
--- a/webrtc/p2p/base/p2ptransportchannel.h
+++ b/webrtc/p2p/base/p2ptransportchannel.h
@@ -232,7 +232,7 @@ class P2PTransportChannel : public TransportChannelImpl,
   void OnConnectionStateChange(Connection* connection);
   void OnReadPacket(Connection *connection, const char *data, size_t len,
                     const rtc::PacketTime& packet_time);
-  void OnSentPacket(const rtc::SentPacket& sent_packet);
+  void OnSentPacket(PortInterface* port, const rtc::SentPacket& sent_packet);
   void OnReadyToSend(Connection* connection);
   void OnConnectionDestroyed(Connection *connection);
 
diff --git a/webrtc/p2p/base/port.cc b/webrtc/p2p/base/port.cc
index 9dd5c83fed..f6a30097f8 100644
--- a/webrtc/p2p/base/port.cc
+++ b/webrtc/p2p/base/port.cc
@@ -310,6 +310,10 @@ void Port::OnReadPacket(
   }
 }
 
+void Port::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  PortInterface::SignalSentPacket(this, sent_packet);
+}
+
 void Port::OnReadyToSend() {
   AddressMap::iterator iter = connections_.begin();
   for (; iter != connections_.end(); ++iter) {
diff --git a/webrtc/p2p/base/port.h b/webrtc/p2p/base/port.h
index 436b1e7faa..57608b5b14 100644
--- a/webrtc/p2p/base/port.h
+++ b/webrtc/p2p/base/port.h
@@ -280,11 +280,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
                             const std::string& remote_ufrag);
 
   // Called when a packet has been sent to the socket.
-  // This is made pure virtual to notify subclasses of Port that they MUST
-  // listen to AsyncPacketSocket::SignalSentPacket and then call
-  // PortInterface::OnSentPacket.
-  virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
-                            const rtc::SentPacket& sent_packet) = 0;
+  void OnSentPacket(const rtc::SentPacket& sent_packet);
 
   // Called when the socket is currently able to send.
   void OnReadyToSend();
diff --git a/webrtc/p2p/base/port_unittest.cc b/webrtc/p2p/base/port_unittest.cc
index 449021ad9f..34806d9a79 100644
--- a/webrtc/p2p/base/port_unittest.cc
+++ b/webrtc/p2p/base/port_unittest.cc
@@ -204,10 +204,6 @@ class TestPort : public Port {
   }
 
  private:
-  void OnSentPacket(rtc::AsyncPacketSocket* socket,
-                    const rtc::SentPacket& sent_packet) {
-    PortInterface::SignalSentPacket(sent_packet);
-  }
   rtc::scoped_ptr<ByteBuffer> last_stun_buf_;
   rtc::scoped_ptr<IceMessage> last_stun_msg_;
   int type_preference_ = 0;
diff --git a/webrtc/p2p/base/portinterface.h b/webrtc/p2p/base/portinterface.h
index e83879f3b7..d1c371d356 100644
--- a/webrtc/p2p/base/portinterface.h
+++ b/webrtc/p2p/base/portinterface.h
@@ -116,7 +116,7 @@ class PortInterface {
                    const rtc::SocketAddress&> SignalReadPacket;
 
   // Emitted each time a packet is sent on this port.
-  sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
+  sigslot::signal2<PortInterface*, const rtc::SentPacket&> SignalSentPacket;
 
   virtual std::string ToString() const = 0;
 
diff --git a/webrtc/p2p/base/relayport.cc b/webrtc/p2p/base/relayport.cc
index 19883a3121..88adcf2f88 100644
--- a/webrtc/p2p/base/relayport.cc
+++ b/webrtc/p2p/base/relayport.cc
@@ -754,7 +754,7 @@ void RelayEntry::OnReadPacket(
 
 void RelayEntry::OnSentPacket(rtc::AsyncPacketSocket* socket,
                               const rtc::SentPacket& sent_packet) {
-  port_->OnSentPacket(socket, sent_packet);
+  port_->OnSentPacket(sent_packet);
 }
 
 void RelayEntry::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
diff --git a/webrtc/p2p/base/relayport.h b/webrtc/p2p/base/relayport.h
index fdbf6e6f98..4b74b91ade 100644
--- a/webrtc/p2p/base/relayport.h
+++ b/webrtc/p2p/base/relayport.h
@@ -29,7 +29,7 @@ class RelayConnection;
 // is created. The RelayEntry will try to reach the remote destination
 // by connecting to all available server addresses in a pre defined
 // order with a small delay in between. When a connection is
-// successful all other connection attempts are aborted.
+// successful all other connection attemts are aborted.
 class RelayPort : public Port {
  public:
   typedef std::pair<rtc::Socket::Option, int> OptionValue;
@@ -96,11 +96,6 @@ class RelayPort : public Port {
                     ProtocolType proto,
                     const rtc::PacketTime& packet_time);
 
-  // The OnSentPacket callback is left empty here since they are handled by
-  // RelayEntry.
-  void OnSentPacket(rtc::AsyncPacketSocket* socket,
-                    const rtc::SentPacket& sent_packet) override {}
-
  private:
   friend class RelayEntry;
 
diff --git a/webrtc/p2p/base/stunport.cc b/webrtc/p2p/base/stunport.cc
index 8f37dd5218..67cf789da8 100644
--- a/webrtc/p2p/base/stunport.cc
+++ b/webrtc/p2p/base/stunport.cc
@@ -340,7 +340,7 @@ void UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
 
 void UDPPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
                            const rtc::SentPacket& sent_packet) {
-  PortInterface::SignalSentPacket(sent_packet);
+  Port::OnSentPacket(sent_packet);
 }
 
 void UDPPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
diff --git a/webrtc/p2p/base/tcpport.cc b/webrtc/p2p/base/tcpport.cc
index 6a64174f00..23afc2f031 100644
--- a/webrtc/p2p/base/tcpport.cc
+++ b/webrtc/p2p/base/tcpport.cc
@@ -258,7 +258,6 @@ void TCPPort::OnNewConnection(rtc::AsyncPacketSocket* socket,
   incoming.socket = new_socket;
   incoming.socket->SignalReadPacket.connect(this, &TCPPort::OnReadPacket);
   incoming.socket->SignalReadyToSend.connect(this, &TCPPort::OnReadyToSend);
-  incoming.socket->SignalSentPacket.connect(this, &TCPPort::OnSentPacket);
 
   LOG_J(LS_VERBOSE, this) << "Accepted connection from "
                           << incoming.addr.ToSensitiveString();
@@ -287,12 +286,6 @@ void TCPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
   Port::OnReadPacket(data, size, remote_addr, PROTO_TCP);
 }
 
-void TCPPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
-                           const rtc::SentPacket& sent_packet) {
-  ASSERT(socket == socket_);
-  PortInterface::SignalSentPacket(sent_packet);
-}
-
 void TCPPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   Port::OnReadyToSend();
 }
diff --git a/webrtc/p2p/base/tcpport.h b/webrtc/p2p/base/tcpport.h
index b298a17e9c..568dc65f3d 100644
--- a/webrtc/p2p/base/tcpport.h
+++ b/webrtc/p2p/base/tcpport.h
@@ -96,9 +96,6 @@ class TCPPort : public Port {
                     const rtc::SocketAddress& remote_addr,
                     const rtc::PacketTime& packet_time);
 
-  void OnSentPacket(rtc::AsyncPacketSocket* socket,
-                    const rtc::SentPacket& sent_packet) override;
-
   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
 
   void OnAddressReady(rtc::AsyncPacketSocket* socket,
diff --git a/webrtc/p2p/base/turnport.cc b/webrtc/p2p/base/turnport.cc
index 5ed93dd1d8..022207aa5b 100644
--- a/webrtc/p2p/base/turnport.cc
+++ b/webrtc/p2p/base/turnport.cc
@@ -351,8 +351,6 @@ bool TurnPort::CreateTurnClientSocket() {
 
   socket_->SignalReadyToSend.connect(this, &TurnPort::OnReadyToSend);
 
-  socket_->SignalSentPacket.connect(this, &TurnPort::OnSentPacket);
-
   // TCP port is ready to send stun requests after the socket is connected,
   // while UDP port is ready to do so once the socket is created.
   if (server_address_.proto == PROTO_TCP) {
@@ -584,11 +582,6 @@ void TurnPort::OnReadPacket(
   }
 }
 
-void TurnPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
-                            const rtc::SentPacket& sent_packet) {
-  PortInterface::SignalSentPacket(sent_packet);
-}
-
 void TurnPort::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
   if (ready()) {
     Port::OnReadyToSend();
diff --git a/webrtc/p2p/base/turnport.h b/webrtc/p2p/base/turnport.h
index 4d83806a37..a19f67632e 100644
--- a/webrtc/p2p/base/turnport.h
+++ b/webrtc/p2p/base/turnport.h
@@ -106,8 +106,6 @@ class TurnPort : public Port {
                             const rtc::SocketAddress& remote_addr,
                             const rtc::PacketTime& packet_time);
 
-  virtual void OnSentPacket(rtc::AsyncPacketSocket* socket,
-                            const rtc::SentPacket& sent_packet);
   virtual void OnReadyToSend(rtc::AsyncPacketSocket* socket);
   virtual bool SupportsProtocol(const std::string& protocol) const {
     // Turn port only connects to UDP candidates.