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Author SHA1 Message Date
mikhal@webrtc.org
3c5a9242fe Don't force cont' when enabling kWithErrors 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2047004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 20:45:36 +00:00
mikhal@webrtc.org
635b2b88e4 Removing some TODO's from libyuv 
BUG=1996
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2146004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 20:06:18 +00:00
mikhal@webrtc.org
2b810bf77b Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2143004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 19:09:49 +00:00
fischman@webrtc.org
ccf8b56670 AppRTCDemo(android): prefer ISAC for audio codec. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2126004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 19:02:11 +00:00
fischman@webrtc.org
8788167b9b PeerConnection Java: explicitly cast DataChannel* to jlong for Java. 
Otherwise on 32-bit ARM Android the nativeDataChannel param the Java ctor sees
is a 64-bit value whose low 32 bits are the pointer, and whose high 32-bits are
garbage.

BUG=2302
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 18:58:12 +00:00
kjellander@webrtc.org
c8c32638be Remove JpegEncoder suppression as jpeg is now removed. 
See https://code.google.com/p/webrtc/source/detail?r=4646

BUG=2322
TEST=Ran common_video_unittests with the suppression removed
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2164004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 18:38:52 +00:00
mflodman@webrtc.org
f5f5da0df1 Adding TSAN suppression for test posix udp transport. 
This is race for reading a bool in the WebRTC test UDP transport and
not in any production code.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 14:15:21 +00:00
kjellander@webrtc.org
3a6ff41e8f Document the source of test scenarios for Dummynet wrapper script. 
I just wanted to put this in here since I got the question
from an external user.

TEST=none
BUG=none
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/2162004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 13:01:31 +00:00
mflodman@webrtc.org
cac7325b84 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. 
Found with tsan.

TEST=try job and tsan
R=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2156004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 12:11:12 +00:00
pbos@webrtc.org
cb5118c14c Add FakeEncoder to VideoSendStream tests. 
Breaks out config part of FakeEncoder from VideoSendStream tests to
FakeEncoder. Also sets FakeEncoder as encoder for VideoSendStream tests.
Anticipated speedup didn't happen as VP8 is still initialized by default
when creating channels in the old API. This will be sped up when moving
off the old API as VP8 won't be enabled by default.

BUG=2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2155004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 09:10:37 +00:00
henrik.lundin@webrtc.org
8fb89533af Correcting two nits in InputAudioFile 
First, the fread function returns number of elements read, not
necessarily the number of bytes. In this case, it is the number
of samples. Second, a spelling mistake was corrected.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2161004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-03 08:43:28 +00:00
mflodman@webrtc.org
8d32066073 Changed method name. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:30 +00:00
mflodman@webrtc.org
814d5e9133 Renamed method. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:18 +00:00
mflodman@webrtc.org
d51bcffc1e Function name change. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:45:09 +00:00
mflodman@webrtc.org
dfbf52baac Fixing capture frame race in ViECapturer. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 12:44:57 +00:00
kjellander@webrtc.org
5aedb295d5 Add TSan and Dr Memory suppressions for Windows 
This works enables us to add add more memory tools bots to the
WebRTC FYI waterfall at
http://build.chromium.org/p/client.webrtc.fyi/waterfall
These suppressions will be needed to get the bots green initially.

This CL also updates the PRESUBMIT.py scripts for the previous
memcheck and TSan suppression directories with the trybots we
currently have. It also adds a PRESUBMIT.py script for the
Dr Memory suppressions.

BUG=1938,2319,2321,2322,2323,2324,2328,2329,2330,2333
TEST=Local execution of the tests passes when these suppressions
are used.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 11:50:46 +00:00
henrik.lundin@webrtc.org
b3e905cd91 Disable all LS_VERBOSE logging in NetEq4 
This reduces exectution time of NetEqDecodingTest.TestBitExactness
with almost 30% and reduces the allocation count (from valgrind)
with almost 50% for the same test.

An issue has been created to re-enable logs when logging performance
is improved; see https://code.google.com/p/webrtc/issues/detail?id=2317.

BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2136004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 09:41:06 +00:00
henrik.lundin@webrtc.org
c487c6abb0 NetEq4: Make the algorithm buffer a member variable 
This reduces the alloc count by more than 100,000 for
NetEqDecodingTest.TestBitExactness.

BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-02 07:59:30 +00:00
wu@webrtc.org
cadf9040cb Update talk to 51664136. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 21:24:16 +00:00
pbos@webrtc.org
a957570d62 Overuse detection based on capture-input jitter. 
This is believed to be more reliable in real-world cases. The camera seems to fall behind sooner than the encoder starts taking too long time encoding, so this is believed to be an earlier trigger.

BUG=2325
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 17:16:32 +00:00
mikhal@webrtc.org
0b960cf7a1 Libjpeg is needed for Libyuv 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 16:30:53 +00:00
mikhal@webrtc.org
cf61bee5a2 Removing JPEG as it is not used. 
BUG= 2322
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2138005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 16:00:08 +00:00
turaj@webrtc.org
45d2840623 Zero comfort noise for stereo insted of assertion. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2084004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 15:37:08 +00:00
turaj@webrtc.org
3170b5750f Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics(). 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 15:36:53 +00:00
sergeyu@chromium.org
9ded07e3a4 Fix typo in InvertedDesktopFrame 
BUG=279334
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2141004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-30 01:05:14 +00:00
kjellander@webrtc.org
bfde359b15 Revert accidental checkin of DEPS 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 13:11:38 +00:00
kjellander@webrtc.org
c520fc9d97 Add svn:ignore on dirs that shouldn't be wiped during gclient revert 
This will speed up our buildbots and trybots since it also enables
incremental builds again (currently out/ was wiped for each build).

The Chromium toolchain somehow has changed behavior, since the out/
dir didn't use to be wiped. This change should get faster builds
back again.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@4641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 13:10:21 +00:00
kjellander@webrtc.org
de499669ef Fix fileutils.cc for tests running under Win memory tools. 
On Windows, when tests are run under memory tools like
DrMemory and TSan, slashes occur in the path as directory
separators. This makes the fileutils.cc fail to figure out
the project root, making paths to resource files invalid.

By replacing slashes in the path with backslashes for Windows
this can be fixed.

BUG=2318
TEST=running the same command as the bot locally, succeeding after this patch is applied.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2137004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 11:26:41 +00:00
kjellander@webrtc.org
f8c16b8bf4 Disabling CondVarTest for TSan v2 (take 2) 
This test goes into an eternal loop sometimes when run under
TSan v2, therefore I'm disabling it until this is resolved.
Doing that, the TSan v2 bot in our new WebRTC FYI waterfall at
http://build.chromium.org/p/client.webrtc.fyi/waterfall
will hopefully go green.

The define used only exists for TSan v2, so v1 should be
unaffected.

The suppression added in http://review.webrtc.org/2126005/
turns out actually has no effect, so I'm removing it again.
It's probably useless since TSan v2 obviously alters the code's
behavior, causing these errors. Since the errors are not TSan v2
failures, the suppression has no effect.

BUG=2259
TEST=test passing locally when run as described in 2259 (since
no tests are run).

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2115005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 09:47:18 +00:00
dwkang@webrtc.org
b295a3f592 Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. 
Background:
Since we had http://review.webrtc.org/2048004, the SSRC value in
RtpRtcp for audio hasn't been updated. Because this prevents NTP update in RtpRtcp, the sync logic in ViESyncModule::Process() does not work.

BUG=b/10484087
TEST= pass 'git try' except tests already broken in http://build.chromium.org/p/tryserver.webrtc/console
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2131004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 07:34:12 +00:00
minyue@webrtc.org
d7301775f5 update neteq 4 to facilitate NACK 
BUG=
R=turaj@webrtc.org, turajs@google.com

Review URL: https://webrtc-codereview.appspot.com/2008004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-29 00:58:14 +00:00
kjellander@webrtc.org
8ae641edfb Add suppressions file for Leak Sanitizer. 
Add a suppressions file that must exist in order for the Leak Sanitizer
(LSan) bot to be able to run with the default configuration.
This bot is located in the new WebRTC FYI waterfall at:
http://build.chromium.org/p/client.webrtc.fyi/waterfall

More info about LSan on http://www.chromium.org/developers/testing/leaksanitizer

BUG=none
TEST=Running a local master+slave passes fine with this file added locally.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2135004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-28 20:29:35 +00:00
kjellander@webrtc.org
5f8d05add2 Disabling CondVarTest for TSan v2. 
This test goes into an eternal loop sometimes when run under
TSan v2, therefore I'm disabling it until this is resolved.
Doing that, our new WebRTC FYI waterfall with a TSan v2 bot will hopefully go green when deployed (see https://codereview.chromium.org/23231004)

The define used only exists for TSan v2, so v1 should be
unaffected.

BUG=2259
TEST=test passing locally when run as described in 2259 (since
no tests are run).

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2126005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-28 19:54:33 +00:00
kjellander@webrtc.org
f746f4f07c Fix metrics_unittests on Android. 
The metrics_unittests tried to write an empty file to the
current working directory, which caused a crash on Android. By
using the output folder instead, this is avoided.
This makes it possible to re-enable the tests disabled in r4474
and also remove the empty dummy test added in r4627.

BUG=1882,
TEST=Setup a Chromium+WebRTC checkout as described in issue 1882
. build/android/envsetup.sh
export GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1"
gclient runhooks
ninja -C out/Release metrics_unittests_apk
build/android/test_runner.py gtest -s metrics_unittests --verbose --release

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2127006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-28 18:31:54 +00:00
pbos@webrtc.org
2b325e8394 TSan suppression for RampUpTest/* and EngineTest/*Nack*. 
BUG=2312
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2132004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-28 10:29:10 +00:00
sergeyu@chromium.org
80b56a71e7 Revert part of libjingle roll that caused flakiness of WebRTC tests. 
BUG=crbug.com/279270
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 22:16:21 +00:00
phoglund@webrtc.org
02421fcf0b Corrected documentation on webrtc_test.sh. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2122004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 14:00:10 +00:00
kjellander@webrtc.org
e141373b8a Add isolate configuration for Android for all tests. 
In https://code.google.com/p/webrtc/source/detail?r=4407
henrike@ added the path to the WebRTC resources and
data directories for Android that are required in order to
use isolate for test execution on Android.

This CL adds similar entries to the rest of the .isolate
files added in
https://code.google.com/p/webrtc/source/detail?r=4590.

It also removes three accidentally added .isolate files that originated
from old test names:
* audio_device_test_api
* video_capture_module_test
* video_render_module_test

BUG=1882,1916
TEST=trybots passing.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 12:10:09 +00:00
tina.legrand@webrtc.org
89502c1ca8 Memory and tsan tests: Turning off renamned tests 
In r4625 AudioCodingModuleTest.RunAllTests was broken down to a number of
smaller tests. This CL turns off these tests, as RunAllTests was turned off
before.

TBR=kjellander@webrtc.org

BUG=issue2173

Review URL: https://webrtc-codereview.appspot.com/2121004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 09:38:10 +00:00
tina.legrand@webrtc.org
ee92b664b3 Re-organizing ACM tests 
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.

While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.

I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().

BUG=issue2173
R=minyue@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1961004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-27 07:33:51 +00:00
elham@webrtc.org
d6fef9d380 Fixing SetDecodeErrorMode build error 
- got introduced when reverting r4562

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2118004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 23:59:38 +00:00
elham@webrtc.org
814e28413d Revert r4562 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2117004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 23:21:03 +00:00
sergeyu@chromium.org
01cb3ad883 Fix image flipping for OpenGL-based screen capturer on Mac. 
I broke captured image flipping when refactoring this code while it was
still in chromium. Previously we had CaptureData that was returned from
capturers with correctly inverted stride, but frames were still stored
with positive stride. CaptureData was removed and so the returned frames
always had positive stride, which is not correct. Now ScreenCapturerMac
uses frames with inverted stride when capturing using OpenGL.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2105004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 21:48:56 +00:00
fischman@webrtc.org
e3de6b1e90 Enable ObjC build by default and reenable 64-bit mac libjingle build 
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2080004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 19:31:21 +00:00
fischman@webrtc.org
4498d013f6 apprtc: rationalize whitespace 
- Remove ^M DOS line endings
- Remove trailing whitespace
- Remove leading 2-space indents from files that have carried this indent since   their contents was removed from within enclosing contexts that required it.
- Add a newline to avoid 82-column line.

R=vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2112004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 18:01:28 +00:00
fischman@webrtc.org
5a035b4279 apprtc: add ctrl+i Info window showing gathered ICE candidate types 
R=vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 17:44:38 +00:00
elham@webrtc.org
6dc45a67ee Updated WebRTC version to 3.40 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2111004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 17:30:54 +00:00
mallinath@webrtc.org
af84d782f0 Initialize ssl_role_ to the default role in FakeTransportChannel 
constructor.
This is needed as BaseSession tests can query the transport channel
without creating dtlstransportchannel ( as they are unaware of the
underlying implementation).

This will also fix the memcheck error in webrtc bots.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2110004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 17:14:13 +00:00
mikhal@webrtc.org
f31a47abdc VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that, 
BUG=
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2077004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 17:10:11 +00:00
phoglund@webrtc.org
c9fa0fede5 Removed build status tracking, refreshed front page. 
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2106004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 08:45:22 +00:00