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Author SHA1 Message Date
henrik.lundin@webrtc.org
e8433eb115 Reimplementing NetEq4's AudioVector 
The current implementation using std::vector is too slow.
This CL introduces a new implementation, using a regular
array as data container.

In AudioMultiVector::ReadInterleavedFromIndex, a special case for
1 channel was implemented, to further reduce runtime. Finally,
AudioMultiVector::Channels was reimplemented.

The changes in this CL reduces the runtime of neteq4_speed_test
by 33%.

BUG=1363
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 13:15:02 +00:00
asapersson@webrtc.org
38599510df Parse next RTCP XR report block after an unsupported block type. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 08:08:26 +00:00
minyue@webrtc.org
3e427263ee Reducing opus_test runtime to pass Android test 
BUG=2609
R=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 22:03:52 +00:00
andrew@webrtc.org
e03cafaebc MIPS optimizations for AECM audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2279005

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 20:10:01 +00:00
andrew@webrtc.org
b0730108a2 Move audio_processing dependencies to a variable. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 17:20:27 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
andrew@webrtc.org
6e908b3adf Remove unnecessary include_dirs from audio_processing. 
TBR=aluebs
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/3659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 19:52:05 +00:00
marpan@webrtc.org
00ed170795 Roll libvpx 225010:232686. 
R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/3649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 18:37:18 +00:00
andrew@webrtc.org
5973f3a24a Remove unneeded includes from trace_posix.cc. 
TESTED=trybots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 17:30:07 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. 
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 15:18:52 +00:00
henrikg@webrtc.org
bff9620116 Fix log build error for Chromium builds. 
This only happens when building in Chromium. Can't roll due to this.

../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc: In function 'Window {anonymous}::GetTopLevelWindow(Display*, Window)':
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: error: 'LS_INFO' was not declared in this scope
../../third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc:39:7: note: suggested alternative:
../../third_party/webrtc/system_wrappers/interface/logging.h:71:29: note:   'webrtc::LS_INFO'

See for example http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20%5Blatest%20WebRTC%2Blibjingle%5D/builds/3039/steps/compile/logs/stdio

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 10:37:27 +00:00
kjellander@webrtc.org
4c828e145e Remove update_resources.py as it's no longer used. 
After http://review.webrtc.org/2095004/ has been landed
for normal WebRTC builds, and https://codereview.chromium.org/62273004/
and https://codereview.chromium.org/60513012/ for our Android
APK builds with a Chromium checkout, we should be fine to remove
this script.

I have verified that the runhooks step on the Android testers
is using the download_from_google_storage.py script to pull
the resources from Google Storage.

BUG=webrtc:2294
TEST=a few trybots passing compile step.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 09:08:36 +00:00
andrew@webrtc.org
f1a48174d4 Replace disabled logging with a restricted logging mode. 
This will enable some low-level webrtc logging in a Chromium build,
while limiting the binary size impact.

For a Mac Release build, it results in an increase to Chrome.app of 37k
and libpeerconnection.so of 25k. For comparison, enabling full logs
costs 230k and 218k respectively.

BUG=b/11470432
TESTED=voe_cmd_test produces logs of the appropriate severity.
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-07 23:47:26 +00:00
elham@webrtc.org
5adc89747a Updated WebRTC version to 3.46 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 22:27:51 +00:00
fbarchard@google.com
a7855a88b3 Fix for xgetbv on Visual Studio 2010. 
BUG=none
TEST=local build of webrtc with 2010.  python build\gyp_chromium --depth=. -G msvs_version=2010 -fninja all.gyp & ninja -C out\Debug
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 21:18:48 +00:00
marpan@webrtc.org
bde3056567 Fix for video_processor_intergration_tests to run in parallel. 
BUG=2601.
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 20:59:29 +00:00
kjellander@webrtc.org
c4225b63bb Update getUserMedia W3C conformance tests. 
This CL updates these tests to the spec as of
http://dev.w3.org/2011/webrtc/editor/archives/20130824/getusermedia.html

There are still a lot of functionality that lacks testing. I've put a bunch of TODOs in there but I'm unlikely to get time to implement them all any time soon...

TEST=local testing with Chrome Canary.
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 13:26:34 +00:00
asapersson@webrtc.org
8bad50e845 Sending status fix for module. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 10:45:58 +00:00
wu@webrtc.org
16d6254e8c Update talk to 56183333. 
TEST=try bots
R=sheu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/3469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 23:45:14 +00:00
kjellander@webrtc.org
d16d307218 Fix bad Google Storage uploads of resource files. 
The files in this CL seem to have hit some kind of bug
during upload, causing the downloaded files to get another
SHA-1 hash than the .sha1 file. This makes them become
redownloaded every time runhooks execute.
Re-uploading them one by one seems to have resolved this.

TEST=trybots passing
BUG=2294
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 21:03:04 +00:00
kjellander@webrtc.org
0e03360591 Add OWNERS for resources/ 
Make it possible for all our committers to
upload resource .sha1 files in here.

TEST=none
BUG=2294
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 21:02:49 +00:00
kjellander@webrtc.org
7a36cb408b Add missing dependencies to .isolate files 
Also fix invalid paths in video_engine_tests.isolate.

TEST=trybots passing compile step (no .isolate use is deployed on them yet)
BUG=chromium:300017
R=pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 14:28:57 +00:00
kjellander@webrtc.org
1e8b671d05 Roll chromium_revision 231713:232627 
This will pick up r232015 which fixes an error during
GYP execution when GYP_DEFINES is not set.

TEST=trybots passing compile step.
BUG=none
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 12:32:05 +00:00
kjellander@webrtc.org
da7f6589aa Add svn:ignore to avoid re-download of resources 
Without this, the bots will download all resources for
every build. This consumes a lot of unnecessary traffic.
I tried experimenting with patterns ignoring everything
except the .sha1 files but wasn't able to get it working,
so this will have to do for now.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 09:27:51 +00:00
fischman@webrtc.org
b8cb85b348 Fix broken build on x86 Android 
BUG=2545
R=fischman@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3019004

Patch from Lu Quiang <qiang.lu@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 19:06:08 +00:00
fischman@webrtc.org
7b273a545d PeerConnection iOS: update README instructions 
This is needed to account for https://codereview.chromium.org/25535004/

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 18:48:12 +00:00
wu@webrtc.org
07a6fbe83d Update talk to 56092586. 
R=jiayl@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 18:41:34 +00:00
kjellander@webrtc.org
3779c1cb0a Fix invalid .sha1 files for audio_coding 
It seems like multiple runs of the upload_to_google_storage.py
script created .sha1.sha1 files that sneaked in with
https://code.google.com/p/webrtc/source/detail?r=5076

This caused the wrong files getting downloaded during sync.
This affected the modules_unittests and the neteq_unittests
which started failing (due to wrong version of the resource files).

TEST=trybots passing
BUG=2294
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 14:54:47 +00:00
kjellander@webrtc.org
80174583bd Replace old resources download script with depot_tools 
With help from hinoka@, we're now using a more efficient approach
to download only the files that have changed from Google Storge.

When uploading new resource files, use
upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename
which of course requires gsutil authentication setup.

NOTICE: Before deploying this, svn:ignore should be removed for
the resources folder, or the bots will run into problems with a
non-versioned file being found in the checkout during sync (as
this CL adds resources to version control).

All developers will also need to be informed to wipe their local
resources dir to avoid getting an error during checkout due to the
already existing non-versioned resources directory.

BUG=2294
TEST=locally running gclient runhooks
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2095004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 12:07:57 +00:00
kjellander@webrtc.org
a452fc29e6 Remove resources/ svn:ignore to prepare for updated resource handling 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 12:07:09 +00:00
kjellander@webrtc.org
58bcdeee2c Roll chromium_revision 229708:231713 
Recent changes in how the build dir is used for bots
(see https://codereview.chromium.org/38873003 for details)
requires us to roll to a more recent version
of Chromium to get our android_apk trybot back into
a working state.

This roll needs to be landed at the same time as the
client.webrtc and tryserver.webrtc masters are updated
with the changes in https://codereview.chromium.org/53283002

TEST=trybots passing (except the iOS ones since they require
the above change to be applied to be able to compile)
BUG=2560
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 09:40:03 +00:00
asapersson@webrtc.org
766154aa1d Removed unused code. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 08:35:50 +00:00
kjellander@webrtc.org
e2df8b7f01 Make video quality analysis unittests print to log instead of stdout. 
I think it's best to avoid printing these perf numbers since
when we turn on perf measurements for Android, it will be for
all tests as far as I understand it works today.

TEST=trybots passing tools_unittests
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-03 18:34:51 +00:00
sheu@chromium.org
5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.

TBR=niklas.emblom@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/3269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 23:41:04 +00:00
sheu@chromium.org
74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:48:16 +00:00
sheu@chromium.org
d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.

Revert while build breakage is fixed.

BUG=None
TBR=niklas.emblom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:20:15 +00:00
sheu@chromium.org
1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 20:32:28 +00:00
wu@webrtc.org
de305014c6 Update talk to 55906045. 
Review URL: https://webrtc-codereview.appspot.com/3159005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:40:38 +00:00
turaj@webrtc.org
58cd31665c Address Clag Analyzer issues. 
Following are the issues related to NetEq 4, discovered by Clang Analyzer.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:15:55 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer. 
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 12:14:34 +00:00
sprang@webrtc.org
da2c37b759 Video bandwidth not reported correctly 
ViEChannel::GetBandwidthUsage fails to aggregate video_bitrate_sent in
the same way as the total, fec and nack.

BUG=2579
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 09:49:03 +00:00
sergeyu@chromium.org
773e72797f Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc 
Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146

BUG=2551
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2759004

Patch from Daniel Nicoara <dnicoara@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 01:51:21 +00:00
wu@webrtc.org
de748c806c Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. 
TEST=build
R=andrew@webrtc.org, fischman@webrtc.org
TBR=andrew

Review URL: https://webrtc-codereview.appspot.com/3149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 20:43:27 +00:00
solenberg@webrtc.org
dce70ccb0b Add delay limit to ChokeFilter. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 19:18:07 +00:00
wu@webrtc.org
f424cb8e13 Update talk to 55863981. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/3089006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 17:57:33 +00:00
solenberg@webrtc.org
d6e46638ec Logging for BWE test framework. 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 16:06:26 +00:00
wu@webrtc.org
cecfd1832d Update talk to 55821645. 
TEST=try bots
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 05:18:12 +00:00
wu@webrtc.org
ec4cccc6b6 Update libyuv to 832. 
R=fbarchard@google.com, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 21:02:20 +00:00
pbos@webrtc.org
47ebbaddbb Make video/ only depend on video_engine_core. 
Fixes Android/Chromium build error. Previous dependencies included
VideoEngine tests that couldn't build on this configuration.

BUG=2535
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 13:11:56 +00:00
pbos@webrtc.org
def22b455b Stop DirectTransports in VideoSendStreamTests. 
Prevents racy packet delivery during or after Call destruction.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 10:12:10 +00:00