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Author SHA1 Message Date
Jon Hjelle
da99da81c9 Update API for Objective-C RTCPeerConnectionFactory. 
BUG=
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1558473002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11326}
 2016-01-20 21:40:35 +00:00
Jon Hjelle
065aacc249 Move RTCVideoSource to webrtc/api/objc. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1546783002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11325}
 2016-01-20 21:25:53 +00:00
danilchap
d8dccd57ea uses standard types instead of RTCPUtility type to store data. 
got member read accessors, got Parse function.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1552773002

Cr-Commit-Position: refs/heads/master@{#11324}
 2016-01-20 20:08:58 +00:00
ivoc
72c08edced Reenables several NetEq unittests on android. 
Several unittests were disabled on android, this CL will reenable them. One of
the tests was accidentally disabled on all platforms, and now no longer gives a
bitexact result.

BUG=webrtc:3343,webrtc:5349

Review URL: https://codereview.webrtc.org/1532903002

Cr-Commit-Position: refs/heads/master@{#11323}
 2016-01-20 15:26:28 +00:00
stefan
32f81542c2 Support REMB in combination with send-side BWE. 
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1581113006

Cr-Commit-Position: refs/heads/master@{#11322}
 2016-01-20 15:14:03 +00:00
Peter Boström
a5dec16b42 Name SimulcastEncoderApdater on InitEncode. 
Provides a better string (provides names of all implementations), but
also fixes a crash when accessing the ImplementationName() of
SimulcastEncoderAdapter where InitEncode has failed.

BUG=chromium:577932, webrtc:4897
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1599353003 .

Cr-Commit-Position: refs/heads/master@{#11321}
 2016-01-20 14:54:02 +00:00
Tommi
9090e0b147 Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding. 
This is a part of cleaning up CriticalSectionWrapper in general.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1610073003 .

Cr-Commit-Position: refs/heads/master@{#11319}
 2016-01-20 12:39:45 +00:00
tommi
84df580d52 Switch to rtc::CriticalSection in IncomingVideoStream and remove one lock. 
BUG=

Review URL: https://codereview.webrtc.org/1608743005

Cr-Commit-Position: refs/heads/master@{#11318}
 2016-01-20 12:38:00 +00:00
Tommi
e8493326f2 Remove ConditionVariableWrapper. 
ConditionVariableEventWin remains for now since it's still needed for the rw lock on Windows XP.

BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1601523009 .

Cr-Commit-Position: refs/heads/master@{#11317}
 2016-01-20 12:36:42 +00:00
tommi
63cb434691 Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/ 
This is a first cl of removing use of CriticalSectionWrapper after a series of cleanup CLs that have been landing recently (and still are landing).

BUG=

Review URL: https://codereview.webrtc.org/1610553002

Cr-Commit-Position: refs/heads/master@{#11316}
 2016-01-20 10:32:58 +00:00
jackychen
f0b8a3784f Allow disabling denoiser when it is enabled. 
BUG=webrtc:5255

Review URL: https://codereview.webrtc.org/1571423003

Cr-Commit-Position: refs/heads/master@{#11312}
 2016-01-20 02:19:01 +00:00
niklas.enbom
3a6bf2d68b Enable full screen windows to be shown in window picker for mac. Before this patch a full screen window can be shared if sharing is started before the window is entered into full screen mode, but not if it's already in full screen. 
BUG=chromium:575990
TEST: Manual test using TextEdit full screen mode.

Review URL: https://codereview.webrtc.org/1579213007

Cr-Commit-Position: refs/heads/master@{#11311}
 2016-01-20 01:34:20 +00:00
Tommi
f01ea4f847 Remove use of ConditionVariableWrapper and CriticalSectionWrapper from UdpSocket2Windows. 
This helps with untangling CriticalSectionWrapper from ConditionVariableWrapper and looks like we can just delete ConditionVariableWrapper and use rtc::Event instead.

BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1606993002 .

Cr-Commit-Position: refs/heads/master@{#11309}
 2016-01-19 21:50:04 +00:00
tommi
cd255cc07b Remove unused ConditionVariableWrapper on POSIX platforms 
BUG=

Review URL: https://codereview.webrtc.org/1602203003

Cr-Commit-Position: refs/heads/master@{#11308}
 2016-01-19 21:13:21 +00:00
Peter Boström
7b971e728b Remove extra_options from VideoCodec. 
Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.

Removes the last webrtc::Config uses/includes from video code.

Also removes VideoCodec equality operators which are no longer in use.

BUG=webrtc:5410
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1606613003 .

Cr-Commit-Position: refs/heads/master@{#11307}
 2016-01-19 15:26:24 +00:00
Tommi
ee5a309f12 Make CriticalSectionWrapper non-virtual. 
There's no need for this class to have a vtable since there exists only a single implementation (per platform).  It's also not good for performance.

BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1601743004 .

Cr-Commit-Position: refs/heads/master@{#11306}
 2016-01-19 14:42:58 +00:00
Peter Boström
dd45eb6801 Remove use-after-free when quality tests stall. 
Reduces TSan warnings when running screenshare FullStack tests.

BUG=
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1601033004 .

Cr-Commit-Position: refs/heads/master@{#11305}
 2016-01-19 14:22:45 +00:00
perkj
8a2c31d208 Make it possible to run peerconnection_unittests on Android. 
- renamed libjingle_peerconnection_unittest to peerconnection_unittest to circumvent cr issue http://crbug.com/543820

TEST=On an android build webrtc/build/android/test_runner.py gtest -s peerconnection_unittests --verbose -t 900
BUG=webrtc:2365,543820
NOTRY=True

Review URL: https://codereview.webrtc.org/1602443004

Cr-Commit-Position: refs/heads/master@{#11304}
 2016-01-19 14:20:07 +00:00
kwiberg
0edb05b344 Declare that rent_a_codec depends on the audio codecs 
That these declarations were missing was a bug, which apparently
didn't actually cause build problems in either Chromium or WebRTC
standalone. (Presumably, because rent_a_codec was always linked
together with other build targets that did declare such dependencies.)

BUG=webrtc:5435

Review URL: https://codereview.webrtc.org/1607463002

Cr-Commit-Position: refs/heads/master@{#11303}
 2016-01-19 13:54:31 +00:00
kjellander
73674f8064 Replace hardcoded constant in video capture with macro. 
The roll in https://codereview.webrtc.org/1593713013 introduced a
cast that is undefined behavior. The right way to fix it is to use
a macro.

NOTRY=True
TESTED=Tommi verified that the values are the same.

Review URL: https://codereview.webrtc.org/1608893003

Cr-Commit-Position: refs/heads/master@{#11302}
 2016-01-19 13:49:25 +00:00
kjellander
3c85cad1d4 Roll chromium_revision 7a4fb8d..f527e86 (370025:370073) 
Change log: 7a4fb8d..f527e86
Full diff: 7a4fb8d..f527e86

No dependencies changed.

Clang was updated 255169:257953.
Details: 7a4fb8d..f527e86/tools/clang/scripts/update.py

NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1593713013

Cr-Commit-Position: refs/heads/master@{#11301}
 2016-01-19 12:47:24 +00:00
tommi
61046eb38d Rename RWLockGeneric to RWLockWinXP to more accurately reflect when it's used. 
Since this is on Windows only, I'm also using the CriticalSectionWrapper and ConditionVariableWrapper Windows types directly which allows us to skip 3 extra heap allocations. It also helps with the removal of the 'friend' relationship between ConditionVariableWrapper and CriticalSectionWrapper, which is causing headaches on Mac.

BUG=

Review URL: https://codereview.webrtc.org/1595983002

Cr-Commit-Position: refs/heads/master@{#11300}
 2016-01-19 11:00:01 +00:00
Peter Boström
8d6fab8fac Remove two dead 'using' instances. 
BUG=
TBR=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1609563003 .

Cr-Commit-Position: refs/heads/master@{#11296}
 2016-01-18 23:24:36 +00:00
Tommi
2067826a5e Remove dependency on ConditionVariableWrapper and CriticalSectionWrapper in UdpSocketPosix. 
This is a part of cleaning up 'friend' parts of ConditionVariableWrapper's implementation where it accesses private variables of CriticalSectionWrapper, which is not good since it makes assumptions about the implementation on all posix platforms.
Instead I'm using rtc::Event, another condition variable based implementation we have, and fits the requirements of UdpSocketPosix.

BUG=
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1591333002 .

Cr-Commit-Position: refs/heads/master@{#11295}
 2016-01-18 19:35:49 +00:00
Peter Boström
233bfd2da4 Move keyframe requests outside encoder mutex. 
Enables faster keyframe requests since they are no longer blocked by
calls to the encoder.

BUG=webrtc:5410
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1600553003 .

Cr-Commit-Position: refs/heads/master@{#11294}
 2016-01-18 19:23:51 +00:00
tommi
aff4b70db0 Simplify the implementation of LoggingTest. 
This removes dependency on ConditionVariableWrapper and CriticalSectionWrapper which currently have a 'friend' relationship that I'd like to get rid of.

BUG=

Review URL: https://codereview.webrtc.org/1590983005

Cr-Commit-Position: refs/heads/master@{#11292}
 2016-01-18 18:20:21 +00:00
kwiberg
f8c2baca4e Add a gyp/gn variable for whether to use iLBC or not 
BUG=webrtc:5415

Review URL: https://codereview.webrtc.org/1578953003

Cr-Commit-Position: refs/heads/master@{#11291}
 2016-01-18 14:38:40 +00:00
danilchap
34ed2b95a5 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function 
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1544983002

Cr-Commit-Position: refs/heads/master@{#11288}
 2016-01-18 10:43:38 +00:00
honghaiz
cec0a08275 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set. 
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.

BUG=

Review URL: https://codereview.webrtc.org/1556743002

Cr-Commit-Position: refs/heads/master@{#11279}
 2016-01-15 22:49:15 +00:00
guoweis
56271ed889 fix bug 5430 
Fixed misusage of Connection function and also fixed the test case.

BUG=webrtc:5430

Review URL: https://codereview.webrtc.org/1592763003

Cr-Commit-Position: refs/heads/master@{#11278}
 2016-01-15 22:45:11 +00:00
deadbeef
884f58523a Storing raw audio sink for default audio track. 
BUG=webrtc:5250

Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
Cr-Commit-Position: refs/heads/master@{#11230}

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11275}
 2016-01-15 17:20:08 +00:00
Danil Chapovalov
1567d0bd98 [rtp_rtcp] rtcp::Sdes moved into own file 
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1439553003/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1592763002 .

Cr-Commit-Position: refs/heads/master@{#11274}
 2016-01-15 16:34:32 +00:00
torbjorng
79a5a83e10 Adapt to boringssl's new defaults. 
This is now a merge with patchset #2 of https://codereview.webrtc.org/1550773002 after that CL was reverted.

BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1589493004

Cr-Commit-Position: refs/heads/master@{#11273}
 2016-01-15 15:16:54 +00:00
Danil Chapovalov
2c13297bf5 [rtp_rtcp] rtcp::Rpsi moved into own file 
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1550293003/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1583233007 .

Cr-Commit-Position: refs/heads/master@{#11272}
 2016-01-15 14:21:34 +00:00
Danil Chapovalov
256e5b23f8 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/ 
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1579213005 .

Cr-Commit-Position: refs/heads/master@{#11271}
 2016-01-15 13:16:36 +00:00
Danil Chapovalov
5679da1291 [rtp_rtcp] rtcp::Fir moved into own file 
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1544403002

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1581983003 .

Cr-Commit-Position: refs/heads/master@{#11269}
 2016-01-15 12:19:59 +00:00
Danil Chapovalov
a5eba6c98b [rtp_rtcp] rtcp::Remb moved into own file 
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1552773002/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1590883002 .

Cr-Commit-Position: refs/heads/master@{#11268}
 2016-01-15 11:40:27 +00:00
ivoc
d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. 
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
 2016-01-15 11:06:41 +00:00
danilchap
d9e62f5837 Fixed sending Rtp packets with non zero csrcs and certain extensions. 
Added test that fails because of given issue.

BUG=webrtc:5413
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1586523003

Cr-Commit-Position: refs/heads/master@{#11258}
 2016-01-14 22:55:23 +00:00
honghaiz
67b1e1ab0b Put options as the argument of the java PeerConnectionFactory constructor. 
BUG=

Review URL: https://codereview.webrtc.org/1581903002

Cr-Commit-Position: refs/heads/master@{#11257}
 2016-01-14 22:45:44 +00:00
terelius
5d332ac8ff Fix expectation bug in the RTPSender unit test. 
The current expectation for InsertPacket(...) uses WillRepeatedly, which accepts if the function is called zero or more times. This CL changes this to either a fixed number of calls, or at least a positive number of calls.

Review URL: https://codereview.webrtc.org/1585783003

Cr-Commit-Position: refs/heads/master@{#11256}
 2016-01-14 22:37:43 +00:00
Stefan Holmer
04cb763955 Add tests for verifying transport feedback for audio and video. 
BUG=webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1589523002 .

Cr-Commit-Position: refs/heads/master@{#11255}
 2016-01-14 19:34:39 +00:00
kjellander
fcfc804e43 Eliminate defines in talk/ 
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).

When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp

NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1588453005

Cr-Commit-Position: refs/heads/master@{#11254}
 2016-01-14 19:01:25 +00:00
sprang
3542013f58 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) 
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.

Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}

TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1586183002

Cr-Commit-Position: refs/heads/master@{#11253}
 2016-01-14 17:14:06 +00:00
Stefan Holmer
2734d77c95 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}
 2016-01-14 16:04:04 +00:00
Stefan Holmer
55674ffb32 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}
 2016-01-14 14:49:23 +00:00
Torbjorn Granlund
31c8d2eac5 Update with new default boringssl no-aes cipher suites. Re-enable tests. 
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).

BUG=webrtc:5381
R=davidben@webrtc.org, henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1550773002 .

Cr-Commit-Position: refs/heads/master@{#11250}
 2016-01-14 14:18:02 +00:00
tommi
e5e0e57bdf 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}
 2016-01-14 12:57:03 +00:00
aluebs
688e308a35 Re-land: "Use an explicit identifier in Config" 
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.

Original CL: https://codereview.webrtc.org/1538643004/

TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1589573004

Cr-Commit-Position: refs/heads/master@{#11248}
 2016-01-14 12:32:51 +00:00
Stefan Holmer
7307952a5b 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}
 2016-01-14 12:15:56 +00:00