History log of /external/webrtc/webrtc/p2p/base/tcpport.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
2734d77c952cfae75fa79fd0631cc34caec4f084 14-Jan-2016 Stefan Holmer <stefan@webrtc.org> Remove assert which was incorrectly added to TcpPort::OnSentPacket.

TBR=pthatcher@webrtc.org

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1588083002 .

Cr-Commit-Position: refs/heads/master@{#11252}
/external/webrtc/webrtc/p2p/base/tcpport.cc
55674ffb32307c6f3efaab442340d3c5c075073b 14-Jan-2016 Stefan Holmer <stefan@webrtc.org> Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.

Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.

R=tommi@webrtc.org
TBR=pthatcher@webtrc.org

BUG=4173

Review URL: https://codereview.webrtc.org/1589563003 .

Cr-Commit-Position: refs/heads/master@{#11251}
/external/webrtc/webrtc/p2p/base/tcpport.cc
e5e0e57bdfd8831b2ad917e7990e273fdfe26af4 14-Jan-2016 tommi <tommi@webrtc.org> 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}
/external/webrtc/webrtc/p2p/base/tcpport.cc
7307952a5bf63311e5f9a2a90089a06177e42716 14-Jan-2016 Stefan Holmer <stefan@webrtc.org> 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

Review URL: https://codereview.webrtc.org/1577873003 .

Cr-Commit-Position: refs/heads/master@{#11247}
/external/webrtc/webrtc/p2p/base/tcpport.cc
f9945b2d1aa2d78b19987219ea872605167d7b5f 15-Dec-2015 Honghai Zhang <honghaiz@webrtc.org> Only try to pair protocol matching candidates for creating connections.
If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516613002 .

Cr-Commit-Position: refs/heads/master@{#11034}
/external/webrtc/webrtc/p2p/base/tcpport.cc
310b093aa41d00be744b6f4bce77fbb657a80096 18-Nov-2015 Guo-wei Shieh <guoweis@webrtc.org> Fix active tcp port to 9

In tcp only call:
Tested with hangout.
Tested with firefox.

To test firefox, goto about:config, search for media.peerconnection.ice.tcp and turn it on.

Existing test case should be suffice to cover this.

R=juberti@google.com
TBR=jubert@webrtc.org
BUG=webrtc:3849

Review URL: https://codereview.webrtc.org/1217463004 .

Cr-Commit-Position: refs/heads/master@{#10683}
/external/webrtc/webrtc/p2p/base/tcpport.cc
0c4e06b4c6107a1b94f764e279e4fb4161e905b0 07-Oct-2015 Peter Boström <pbos@webrtc.org> Use suffixed {uint,int}{8,16,32,64}_t types.

Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/webrtc/p2p/base/tcpport.cc
53eee43e7873d5de2ae015b48c1c353dc0a32d92 23-Sep-2015 Guo-wei Shieh <guoweis@webrtc.org> Address the comment from 1367553002.

Remove duplication introduced by
https://codereview.webrtc.org/1367553002

BUG=webrtc:5030
TBR=juberti@webrtc.org

Review URL: https://codereview.webrtc.org/1360203003 .

Cr-Commit-Position: refs/heads/master@{#10039}
/external/webrtc/webrtc/p2p/base/tcpport.cc
2e4b620471efdd913e2d9975b9d0b49b3e50f3a7 23-Sep-2015 Guo-wei Shieh <guoweis@webrtc.org> TcpPort doesn't connect when calling gmail with non-proxied UDP disabled.

The same check has been made into turnport.cc but missed this place.

BUG=webrtc:5030
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1367553002 .

Cr-Commit-Position: refs/heads/master@{#10038}
/external/webrtc/webrtc/p2p/base/tcpport.cc
6304626268238a074051910d201e9a77aae677e0 14-Sep-2015 Tim Psiaki <tpsiaki@google.com> Add a rate tracker that tracks rate over a given interval split up into buckets that accumulate unit counts for their portion of said interval and use this instead of the standard rate tracker so that the values of retrieved frame rate stats are completely independent of the polling rate.

BUG=
R=asapersson@webrtc.org, noahric@chromium.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1279433006 .

Cr-Commit-Position: refs/heads/master@{#9933}
/external/webrtc/webrtc/p2p/base/tcpport.cc
1eb87c7d94916a6e5641bfbbc8df37527712e4ca 25-Aug-2015 Guo-wei Shieh <guoweis@webrtc.org> TCPConnection can never be deteted if they fail to connect.

Since the TCPConnection has never been connected, they are not scheduled for ping hence will never be detected.

Also fix the case when reconnect fails, as it has become READABLE before, it also will not be deleted.

BUG=webrtc:4936
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1307083002 .

Cr-Commit-Position: refs/heads/master@{#9782}
/external/webrtc/webrtc/p2p/base/tcpport.cc
b594041ec8a3ae9f501260e2456d9d5ce6482819 24-Aug-2015 Guo-wei Shieh <guoweis@webrtc.org> TcpPort Reconnect should inform upper layer to start sending again.

During the reconnection phase, EWOULDBLOCK has been returned to upper layer which stops the sending of video stream.

BUG=webrtc:4930
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1288553010 .

Cr-Commit-Position: refs/heads/master@{#9767}
/external/webrtc/webrtc/p2p/base/tcpport.cc
3d564c10157d7de1d2d4236f4e2a13ff1363d52b 20-Aug-2015 Guo-wei Shieh <guoweis@webrtc.org> Add instrumentation to track the IceEndpointType.

The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.

BUG=webrtc:4918
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1277263002 .

Cr-Commit-Position: refs/heads/master@{#9737}
/external/webrtc/webrtc/p2p/base/tcpport.cc
be508a1d3634ce63b64cd740c44600453e3c3a6b 06-Apr-2015 Guo-wei Shieh <guoweis@chromium.org> Implement Tcp Reconnect for TCPPort.

UDP case should not be changed.

Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.

The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.

Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.

BUG=1926
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31359004

Cr-Commit-Position: refs/heads/master@{#8929}
/external/webrtc/webrtc/p2p/base/tcpport.cc
930e004a817ed346a99ac8e56575326ca75e72aa 17-Nov-2014 guoweis@webrtc.org <guoweis@webrtc.org> Add jmi field for packets discarded due to network error

Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7693

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
6a782c2a46d83e09bb036d34b8c2363adc26d037 14-Nov-2014 henrike@webrtc.org <henrike@webrtc.org> Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.

TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
312614a438c2104ccab6d0231d17604359674e15 13-Nov-2014 guoweis@webrtc.org <guoweis@webrtc.org> Add jmi field for packets discarded due to network error

Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
43e033e7785fff6815fb2154cee87af893fe47a4 10-Nov-2014 henrike@webrtc.org <henrike@webrtc.org> Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted."

BUG=3379
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
332331fb01f8a316ac6d61cf4572478610fb3472 06-Nov-2014 pkasting@chromium.org <pkasting@chromium.org> Use uint16s for port numbers in webrtc/p2p/base.

This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.

This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).

BUG=chromium:81439
TEST=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
269fb4bc90b79bebbb8311da0110ccd6803fd0a8 28-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
28100cb38896fe298b6df11ffd31838d9faf5b8a 18-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."

BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc
d1ba6d9cbfc44618d2c553ff7851948c730ae37b 15-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/p2p/base/tcpport.cc