Commit Graph

5005 Commits

Author SHA1 Message Date
hta@webrtc.org
1009798b31 Demo of multi-pass encode - used for testing limits.
This demo creates a sequence of PeerConnections, and passes
a videostream through all of them.
This allows one to test how many PeerConnections and how
many encodes/decodes the implementation will support before
breaking down or slowing down enough to be unusable.

BUG=
R=fischman@webrtc.org, hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5557 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 06:13:41 +00:00
andrew@webrtc.org
f92aaff104 AudioProcessing is not a Module.
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/8779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 04:22:49 +00:00
henrike@webrtc.org
b8c254abd6 (Auto)update libjingle 61549749-> 61608469
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:38:45 +00:00
bjornv@webrtc.org
e2fc13e42f Refactoring common_audio/signal_processing: Removed two macros used by isac only.
Removed a macro for malloc() and one for free().  They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.

BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org, turajs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
fischman@webrtc.org
c5d506a106 AppRTCDemo(android): clarified README on how to launch app using adb.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 17:55:13 +00:00
stefan@webrtc.org
505f2a0348 Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck.
BUG=2924
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 12:38:06 +00:00
stefan@webrtc.org
9075d519a2 Adding a critical section missing in r5543.
This fixes a race caught by the linux tsan bot.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 09:45:58 +00:00
fischman@webrtc.org
a3708ecdfe PeerConnectionTest(java): unbreak following 61460797-p10
BUG=1414
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 01:51:33 +00:00
mallinath@webrtc.org
385857dfd4 Update talk to 61549749.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 00:56:12 +00:00
wu@webrtc.org
b9a088b920 Update talk to 61538839.
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/8669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 23:18:49 +00:00
wu@webrtc.org
0de29504ab Revert 5545 "Update libjingle to 61514460"
> Update libjingle to 61514460
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8649004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 19:54:28 +00:00
andrew@webrtc.org
38bf249049 Initialize output_will_be_muted_.
TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 17:43:44 +00:00
xians@webrtc.org
e749c9ebdb Update libjingle to 61514460
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 15:09:40 +00:00
asapersson@webrtc.org
8f690bc222 Increase overuse and normal use thresholds for Mac.
BUG=1577
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 14:43:18 +00:00
stefan@webrtc.org
ae2563ae2f Fixes a race when writing to send_padding_.
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 13:48:38 +00:00
kjellander@webrtc.org
12cb88cab9 Add check to verify tree is open to PRESUBMIT.py.
This will disallow commits when our tree is closed.

BUG=chromium:342743
TEST=ran git cl presubmit with an open tree (no error). Then I closed the tree at http://webrtc-status.appspot.com and ran it again, got this message:
Tree state is: closed

***************
Tree is temporarily closed (testing presubmit hook real quick)
http://webrtc-status.appspot.com/current?format=json
***************

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:53:43 +00:00
henrik.lundin@webrtc.org
fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.

This change is only made for NetEq4.

R=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/8599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:42:28 +00:00
fischman@webrtc.org
3eda643a91 PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
BUG=2912
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 04:01:04 +00:00
fischman@webrtc.org
540acde5b3 PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.

Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
  screen off)

BUG=2575
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/8269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 03:56:14 +00:00
andrew@webrtc.org
17342e5092 Add a method to inform AudioProcessing that its output will be muted.
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 22:28:31 +00:00
jiayl@webrtc.org
de782180b0 Change the type of propagation delta from int64 to int.
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.

BUG=2910
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
andrew@webrtc.org
07b5950c12 Initialize key_pressed_.
Was resulting in an error on Mac Asan:
[ RUN      ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():

TBR=aluebs

Review URL: https://webrtc-codereview.appspot.com/8539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 16:41:13 +00:00
andrew@webrtc.org
ce8e077cf0 Add a keypress field to the audioproc debug proto.
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.

TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.

R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 15:28:30 +00:00
pbos@webrtc.org
8118f1861f Set pacing bitrates in SetEncoder.
Before the change no padding was allowed before the first remote bitrate
estimation was received. This bitrate estimation is based on what's
actually sent. In tests I set codec.startBitrate to 300 instead of
325, which incidentally means that only the first layer gets encoded.
As we only send ~150kbps instead of 300, the first REMB will
significantly pull down the remote bitrate estimate instead of keeping
the existing rate, even though there's no problem with the link.

This was detected in RampUpTest.PacingWithRtx,
(send_config.codec.startBitrate=300), which caused the tests to become a
lot slower, and flake out more. By allowing padding initially we're able
to keep our initial bitrate estimate.

R=stefan@webrtc.org
TEST=Running RampUpTest.WithPacingAndRtx with startBandwidth=300.
BUG=

Review URL: https://webrtc-codereview.appspot.com/8529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 14:50:29 +00:00
solenberg@webrtc.org
67e70442b5 Remove unused and not working voe_extended_test.
BUG=2913
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5533 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:58:49 +00:00
pbos@webrtc.org
5591046ab1 .gitignore: + /third_party/{clang_format,usrcsctp}
clang_format and usrcsctp are both synced in through gclient and should
be suppressed.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5532 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:33:22 +00:00
jiayl@webrtc.org
14d80793a8 PeerConnectionClient needs to initialize SSL.
BUG=2911
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 00:41:59 +00:00
andrew@webrtc.org
b659e2844d Reduce mixing threshold in test to avoid flakiness.
Flake observed here:
http://chromegw/i/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/953/steps/voe_auto_test/logs/stdio

TBR=andresp

Review URL: https://webrtc-codereview.appspot.com/8489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5530 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 21:52:50 +00:00
andrew@webrtc.org
75dd2885c5 Add an interface for accepting keypress signals to AudioProcessing.
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5529 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 20:52:30 +00:00
andrew@webrtc.org
aa1278de46 Rename merged webrtc lib to libwebrtc_merged.a.
The name "libwebrtc.a" was conflicting with the newish webrtc target,
resulting in this error:
$ ./webrtc/build/gyp_webrtc merged_lib.gyp
$ ninja -C out/Debug
ninja: warning: multiple rules generate libwebrtc.a. builds involving
this target will not be correct; continuing anyway
ninja: error: dependency cycle: no_op -> libwebrtc.a -> no_op

BUG=b/12955740
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 18:22:29 +00:00
fischman@webrtc.org
8685af7ea0 Remove "Too long processing time of Incoming frame" logspam.
This isn't indicative of anything actionable and spams android logcat with times
in the 10-30ms range several times per second.

BUG=2732
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 17:48:11 +00:00
turaj@webrtc.org
a80be4b23c Add boundary checking to supress gcc 4.8.3 warning.
BUG=2888
Test=try, voe_cmd_test

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
solenberg@webrtc.org
fc320466d1 Remove ViE external encryption API.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 15:27:49 +00:00
michaelbai@google.com
82ebb463fd Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Committed: https://code.google.com/p/webrtc/source/detail?r=5517

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 04:48:27 +00:00
wjia@webrtc.org
dd82fa726c Revert 5516 "Thread annotation of talk_base::CriticalSection."
r5516 failed compilation on builds with enable_webrtc=0.

> Thread annotation of talk_base::CriticalSection.
> 
> Also enabling -Wthread-safety in talk/build/common.gypi for clang on
> Linux. Thread annotations are compile-time checks that for instance
> certain locks are held before accessing a value.
> 
> BUG=
> TEST=Local GUARDED_BY() annotations.
> R=andresp@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8189004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 23:20:15 +00:00
andrew@webrtc.org
16c08f03da Restore mixing integration tests.
These high level tests were disabled over time. Since they depend on
real-time results and the filesystem, they tended to be flaky on the
bots. We now give it a very generous 1 second to start up all channels
before verification and a further relaxed file length check. If we
continue to see problems, I will up the startup delay.

The restored tests would have caught the AGC bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5454

Add a new "real audio" stress test to exercise more code paths. This
would have caught the refactor bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5437

BUG=2164,2844
TESTED=git try. Verified it would have caught the aforementioned bugs
by reintroducing them.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 23:04:39 +00:00
kjellander@webrtc.org
c68d046bb1 Fix BUILD.gn to load all Chromium GN configurations.
After troubleshooting with brettw@chromium.org we
found that this line is needed in order to get the
build/config/BUILD.gn configurations loaded.

This should solve the runhooks failures in
http://build.chromium.org/p/client.webrtc.fyi/waterfall
and make it possible to generate projects with
GYP_DEFINES="disable_glibcxx_debug=1"
(which fails today).

TEST=Ran on Linux:
GYP_DEFINES=build_with_tool=tsan gclient runhooks
without and with this patch applied (it fails without).

R=andrew@webrtc.org, brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5521 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 21:28:55 +00:00
michaelbai@google.com
a65abf9d3a Revert "Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file."
This reverts commit 7686f0ddda.

BUG=

TBR=andrew@webrtc.org, fischman@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/8369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5520 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:26:26 +00:00
jiayl@webrtc.org
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation.
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00
fischman@webrtc.org
82387e4608 Add ability to receive calls for iOS
BUG=2701
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7989005

Patch from Sajid Hussain <shussain@temasys.com.sg>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 18:47:11 +00:00
michaelbai@google.com
7686f0ddda Use libvpx's obj_int_extract and unpack_lib_posix to generate offset header file.
This patch removes generate_asm_header.gypi and uses libvpx's obj_int_extract and
unpack_lib_posix to generate offset header files.

It make the simliar feature's implementation consistent.

R=andrew@webrtc.org, fischman@webrtc.org, fischman@chromium.org
BUG=334447

Review URL: https://webrtc-codereview.appspot.com/7769006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5517 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 17:42:34 +00:00
pbos@webrtc.org
0a7085ffc2 Thread annotation of talk_base::CriticalSection.
Also enabling -Wthread-safety in talk/build/common.gypi for clang on
Linux. Thread annotations are compile-time checks that for instance
certain locks are held before accessing a value.

BUG=
TEST=Local GUARDED_BY() annotations.
R=andresp@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5516 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 13:58:37 +00:00
kjellander@webrtc.org
9cba2e4802 Exclude libjingle_p2p_unittest tests failing TSan Race verifier.
These tests have been failing for a long time at
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20TsanRV
This bot runs a tool called TSan Race Verifier.
More info about Race Verifiers can be found at
https://code.google.com/p/data-race-test/wiki/RaceVerifier

By excluding the tests like this, they will also be disabled
for the normal TSan bot, where they currently pass. This shouldn't
be a problem since we have a TSan v2 bot running too (which uses
a different set of suppressions), so it should catch any newly
introduced race conditions. The TSan v2 bot will soon replace the
old TSan bot, as v1 is being deprecated.

TBR=mallinath@webrtc.org
BUG=2396
TEST=Passing run on Linux of:
GYP_DEFINES=build_with_tool=tsan gclient runhooks
ninja -C out/Release libjingle_p2p_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan_rv -b out/Release -t libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/8329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5515 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 12:43:40 +00:00
sprang@webrtc.org
6f0ca57fb2 Add experiment: SkipEncodingUnusedStreams
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 09:20:51 +00:00
kjellander@webrtc.org
4723dc88b3 Revert 5511 "Revert 5510 "Disable failing libjingle_p2p_unittest..."
So, the test apparently failed right away at 

http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/1224/steps/libjingle_p2p_unittest/logs/stdio


> Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
> 
> According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
> r5505 was committed to resolve exactly these flakes.
> Let's revert the disabling and see.
> 
> BUG=2907
> TBR=mallinath@webrtc.org
> 
> > Disable failing libjingle_p2p_unittest test on Linux
> > 
> > I realize this diables 84 test cases and for all platforms, which
> > I'm not really comfortable with. I tried finding a better way but
> > couldn't without doing significant changes to the file.
> > I think the tests either needs to be fixed or otherwise refactored
> > in order to make more fine-grained disabling possible.
> > 
> > Another (too) large disabling was done by holmer@ in
> > https://webrtc-codereview.appspot.com/2227004 where he should only have
> > disabled them on Windows, if the failures in webrtc:2383 was all that
> > caused those flakes.
> > 
> > BUG=2907
> > TEST=Verified this ran 0 tests:
> > out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> > TBR=wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/8309004
> 
> TBR=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8329004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-09 18:48:56 +00:00
kjellander@webrtc.org
607c805b87 Roll chromium_revision 245382:249215
The find_depot_tools.py is needed to workaround the import
error we get from gyp_chromium when importing it in
webrtc/build/gyp_webrtc (to avoid code duplication).
gyp_chromium introduced a dependency on it in
http://crrev.com/245412 but as we cannot sync all of Chrome's
src/tools (it's quite big), we'll work around this by
adding an empty find_depot_tools module.

The removal of the Cygwin relates to
http://crrev.com/248802 which is a step on the way to remove
Cygwin in Chromium. We seem to already be able to remove it
entirely for WebRTC though.

Changes in the isolate framework required us to update our
copies of the isolate.gypi files.

BUG=none
TEST=trybots passing on all platforms
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-09 18:38:31 +00:00
kjellander@webrtc.org
ce2b44532e Revert 5510 "Disable failing libjingle_p2p_unittest test on Linux"
According to https://code.google.com/p/webrtc/issues/detail?id=2907#c2
r5505 was committed to resolve exactly these flakes.
Let's revert the disabling and see.

BUG=2907
TBR=mallinath@webrtc.org

> Disable failing libjingle_p2p_unittest test on Linux
> 
> I realize this diables 84 test cases and for all platforms, which
> I'm not really comfortable with. I tried finding a better way but
> couldn't without doing significant changes to the file.
> I think the tests either needs to be fixed or otherwise refactored
> in order to make more fine-grained disabling possible.
> 
> Another (too) large disabling was done by holmer@ in
> https://webrtc-codereview.appspot.com/2227004 where he should only have
> disabled them on Windows, if the failures in webrtc:2383 was all that
> caused those flakes.
> 
> BUG=2907
> TEST=Verified this ran 0 tests:
> out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
> TBR=wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/8309004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-09 18:21:00 +00:00
kjellander@webrtc.org
8d2ddd00f1 Disable failing libjingle_p2p_unittest test on Linux
I realize this diables 84 test cases and for all platforms, which
I'm not really comfortable with. I tried finding a better way but
couldn't without doing significant changes to the file.
I think the tests either needs to be fixed or otherwise refactored
in order to make more fine-grained disabling possible.

Another (too) large disabling was done by holmer@ in
https://webrtc-codereview.appspot.com/2227004 where he should only have
disabled them on Windows, if the failures in webrtc:2383 was all that
caused those flakes.

BUG=2907
TEST=Verified this ran 0 tests:
out/Release/libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest.TestNAT_ADDR_RESTRICTEDToNAT_PORT_RESTRICTEDAsGiceBothSharedUfragWithMinimumStepDelay
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5510 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 21:35:20 +00:00
kjellander@webrtc.org
6f8acdc76f Suppressions for libjingle_unittest after roll in r5502
New errors arrived when rolling libjingle in r5502.
These suppressions are needed to green up the Memcheck and
TSan bots.

BUG=1976,2080
TEST=local runs on Linux:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -b out/Release -t libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool=memcheck -b out/Release -t libjingle_unittest
and trybot:
git try --bot=linux_memcheck,linux_tsan -t libjingle_unittest
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5509 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 21:15:59 +00:00
sergeyu@chromium.org
cc685acbdf Disable AsyncInvokeTest.CancelInvoker test
Test is flaky.

BUG=b/12944358
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 19:59:00 +00:00