admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
11,877 Commits 1 Branch 0 Tags
Commit Graph

142 Commits

Author SHA1 Message Date
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
houssainy@google.com
39b1743116 Adding the subtool rtcBot report visualizer 
This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 09:26:16 +00:00
houssainy@google.com
0d3e254c89 Adding three video conference bots test 
A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 16:45:07 +00:00
houssainy@google.com
0e19d0c2aa Adding file from test.webrtc.org domain to be downloaded 
This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 15:41:30 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms. 
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:57:56 +00:00
houssainy@google.com
2192701135 Using the Unused turn configuration in two way test 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 08:40:53 +00:00
houssainy@google.com
fce8f5d319 NOTE: This code review based on the running issue: 
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:24:20 +00:00
houssainy@google.com
3382059e55 Adding Two way video and audio streaming test to RtcBot 
NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:17:15 +00:00
houssainy@google.com
e9b7d03db6 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. 
This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 16:34:25 +00:00
houssainy@google.com
0371a37f85 Moving creating TURN configration to the host machine instead of the bots - rtcBot 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:43:50 +00:00
houssainy@google.com
c221db6165 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. 
Because the symbol ">"  is interpreted as special command for output to file in bash commands.

TBR= andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 09:13:43 +00:00
andresp@webrtc.org
458c2c3b06 Improve rtcbot to load all test files at start and allow them to registerTests 
via: registerBotTest. After loading all tests main.js starts running the
requested one on the command arguments.

R=houssainy@google.com

Review URL: https://webrtc-codereview.appspot.com/29779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 07:36:37 +00:00
houssainy@google.com
3e2f8ff36c Selecting bot_type changed to be specified in the test file 
Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.

Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 15:01:11 +00:00
houssainy@google.com
5d3e7ac1a3 Check on the existence of report directory 
Reports will be written at rtcBot/test/reports/<report_name>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:21:27 +00:00
kjellander@webrtc.org
3037bc3447 GN: Add common configs to tools and test. 
Similar changes as in https://review.webrtc.org/28589004/
were missed in https://review.webrtc.org/25569004/.
This should fix the Chromium WebRTC FYI bots that currently
are broken due to lack of include paths.

BUG=3441
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 19:07:58 +00:00
kjellander@webrtc.org
b8caf6a504 GN: Enable libvpx, add link target and convert some test targets 
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).

I also converted a few test targets and made a GN file for
third_party/gflags.

BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 18:05:02 +00:00
houssainy@google.com
d0bb5862f5 Collecting stats every fixed time in webrtc_video_streaming.js test 
and prepare the format these collected stats to be plotted using one of
external dev-tools.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:20:15 +00:00
kjellander@webrtc.org
e794c36637 Fix parallel test execution for tools, testsupport and metrics tests. 
BUG=2600
TESTED=Passing tests using:
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/test_support_unittests
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/tools_unittests
python third_party/gtest-parallel/gtest-parallel -w 10 -r 20 out/Release/video_engine_tests

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 11:47:28 +00:00
houssainy@google.com
24f62e1a28 Adding getStats function to the exposed PeerConnection in RtcBot 
Exposed Peerconnection object has new function "getStats". This function
returns the stats as array of reports, and each report is RTCStatReport
with additional attributes names and stats.

names: array of all the stat names in current report.
Stats: dictionary and the key is the stat name, and value is the value
of this stat.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 09:36:28 +00:00
houssainy@google.com
35850ff71f Adding test file path as argument of the rtcBot run command's arguments. 
The new command to run rtcBot is:-
node test.js <bot_type> <test_file_path>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 15:24:56 +00:00
houssainy@google.com
07ca949346 Adding webrtc_video_streaming test 
This test is streaming video and audio between two bots using webrtc js api.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 13:52:39 +00:00
houssainy@google.com
e0761d06b0 Fix printing of error stack in rtcbot when a test fails via test.fail(). 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 14:35:35 +00:00
houssainy@google.com
04b853b56a Bot Browser files moved to /bot/browser/ 
because android files will be a different and will need to add more files for Android.

There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:50:09 +00:00
houssainy@google.com
9600519147 Adding the ability to test on Chrome for Android. 
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:01:40 +00:00
houssainy@google.com
c77e4d6aef - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. 
- Select BotType using nodeJs terminal command.

- ping_pong.js test added.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 10:36:11 +00:00
houssainy@google.com
38ef664418 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between 
/borwser and /android

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:44:47 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths. 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 09:55:40 +00:00
kjellander@webrtc.org
0372b93118 Partial revert of r7014 (Android APK refactor) 
This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 14:34:46 +00:00
andresp@webrtc.org
468516c959 RTCBot is a framework that allows to write tests where logic runs on a single 
host that controls multiple endpoints ("bots"). Thus allowing to create more
complex scenarios that would otherwise require non-trival signalling between
multiple parties.

R=houssainy@google.com, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 10:52:54 +00:00
kjellander@webrtc.org
3bd4156d75 Android APK tests built from a normal WebRTC checkout. 
Restructure how the Android APK tests are compiled now
that we have a Chromium checkout available (since r6938).

This removes the need of several hacks that were needed when
building these targets from inside a Chromium checkout.
By creating a symlink to Chromium's base we can compile the required
targets. This also removes the need of the previously precompiled
binaries we keep in /deps/tools/android at Google code.

All the user needs to do is to add the target_os = ["android"]
entry to his .gclient as described at
https://code.google.com/p/chromium/wiki/AndroidBuildInstructions

Before committing this CL, the Android APK buildbots will need
to be updated.
This also solves http://crbug.com/402594 since the apply_svn_patch.py
usage will be similar to the other standalone bots.
It also solves http://crbug.com/399297

BUG=chromium:399297, chromium:402594
TESTED=Locally compiled all APK targets by running:
GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release

checkdeps

R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-01 11:06:37 +00:00
kjellander@webrtc.org
a1bfc50a72 Pass GYP DEPTH variable to isolate. 
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.

Also update all our .isolate files to use the <(DEPTH)
variable.

BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.

R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-13 09:02:15 +00:00
kjellander@webrtc.org
7b82c18979 Add kjellander@webrtc.org as OWNER for *.isolate 
This should make project-wide changes for isolate files
easier and make it more obvious who's a suitable reviewer
for them.

BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-10 05:42:53 +00:00
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions. 
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 20:46:50 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 21:18:46 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..." 
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 11:07:29 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:20:44 +00:00
andresp@webrtc.org
a36ad6929d Add webrtc field trials API. 
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 12:24:04 +00:00
andrew@webrtc.org
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>. 
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:10:28 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:08:03 +00:00
andresp@webrtc.org
e6013bb7be Fix loopback test for case where no constraint is given. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:21:45 +00:00
andresp@webrtc.org
0273fa98e0 Add ability to control peer connection constraints for the loopback test. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 09:40:16 +00:00
andresp@webrtc.org
5a0218c794 Allow loopback tests to do TURN when served from webrtc.googlecode.com. 
BUG=3037
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 19:24:45 +00:00
andresp@webrtc.org
44eb87e6dd Tool to establish a loopback call via apprtc turn server. 
For now the test keeps track of video bandwidth estimation and plots it
using google visualization libraries after the test is concluded.
There is also scripts to run a test and record the tcpdump.

BUG=3037
R=hta@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-17 14:23:22 +00:00
mcasas@webrtc.org
4375e1acbb Add support for YUV4MPEG file reading to tools files. (Minor fix). 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-14 12:51:02 +00:00
mcasas@webrtc.org
6e2d012b58 Add support for YUV4MPEG file reading to tools files. 
This CL adds support for reading .y4m files to the infra in
video_quality_analysis.cc, adding new functions 
ExtractFrameFromYuvFile() and ExtractFrameFromY4mFile(),
instad of the previous ExtractFrameFromI420(). The decision
as to which one to use is taken from the file extension,
if it is .y4m then is considered a YUV4MPEG file, otherwise
is taken as a raw .yuv file.

It also removes the pseudo duplicated function 
GetNextI420Frame(), that is used from psnr_ssim_analyzer.c,
and adds support for y4m files there.

Tested/validated via local compile-run.

YUV4MPEG is a trivial container with a file header
and a per-frame header, see [1]

[1]
http://wiki.multimedia.cx/index.php?title=YUV4MPEG2

BUG=https://code.google.com/p/chromium/issues/detail?id=343504



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-14 12:45:45 +00:00
phoglund@webrtc.org
cf9d364063 Now printing less output from compare_videos.py. 
Alternative solution to the one in
https://codereview.chromium.org/114003006/.

I considered adding a verbose flag, but it needs to be passed through
like 5 functions, so I didn't think it was worth it for a function of
such speculative use.

BUG=chromium:327990
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 17:59:30 +00:00
kjellander@webrtc.org
f9bdbe3619 Roll chromium_revision 232627:238260 
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 13:37:12 +00:00
phoglund@webrtc.org
3260f109e3 Made video quality toolchain more configurable. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 14:10:20 +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
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