88 Commits

Author SHA1 Message Date
andrew@webrtc.org
63e0964039 Fix webrtc compilation errors for Chrome Win64
Mostly disabling warnings in the gyp files.

BUG=1348
BUG=http://crbug.com/166496
BUG=http://crbug.com/167187

Review URL: https://webrtc-codereview.appspot.com/1063011
Patch from Justin Schuh <jschuh@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3423 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 06:45:22 +00:00
mflodman@webrtc.org
59d209562f Moving ViE test files and deleting files no longer used.
BUG=977
TEST=Try bots.

Review URL: https://webrtc-codereview.appspot.com/1046004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3414 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-25 12:45:39 +00:00
elham@webrtc.org
a812a3acee Updated version number to 3.21
Review URL: https://webrtc-codereview.appspot.com/1068004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3399 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 19:39:45 +00:00
wjia@webrtc.org
a3c82bf667 Remove '<(library)' in gyp files.
This will remove all usage of '<(library)' in all webrtc gyp files. 
Review URL: https://webrtc-codereview.appspot.com/1049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18 23:42:21 +00:00
stefan@webrtc.org
3b7feb2a5d Convert psnr and ssim to strings before printing them.
Review URL: https://webrtc-codereview.appspot.com/1042004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3380 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 13:35:01 +00:00
mikhal@webrtc.org
2fd947fb21 Removing outdated comment
Review URL: https://webrtc-codereview.appspot.com/1025007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3376 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-16 18:50:35 +00:00
stefan@webrtc.org
77a584be1d Made ViEToFileRenderer use a separate thread for rendering frames to file.
Review URL: https://webrtc-codereview.appspot.com/1021011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3373 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-15 16:34:34 +00:00
braveyao@webrtc.org
49273ffa79 logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid()
BUG = Issue1283
Review URL: https://webrtc-codereview.appspot.com/1013008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3371 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 01:52:26 +00:00
stefan@webrtc.org
e7dc7f8553 Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM.
TBR=mflodman

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1032005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3360 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-11 12:55:19 +00:00
stefan@webrtc.org
e468f08078 Disable PSNR/SSIM thresholds for the Gilber-Elliot test.
This is to avoid flakiness as the GE model can cause quite big freezes
from time to time. Will keep the test running to get the plots.

TBR=phoglund

BUG=1271

Review URL: https://webrtc-codereview.appspot.com/1030004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-10 15:17:36 +00:00
phoglund@webrtc.org
d005468e9b Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac.
BUG=1268
TEST=vie_auto_test on mac and linux
TBR=mflodman, kjellander

Review URL: https://webrtc-codereview.appspot.com/1027006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3347 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 16:53:42 +00:00
henrika@webrtc.org
d66eb8c4eb Disabled GQoS since it breaks ViE auto test.
BUG=1266
TEST=vie_auto_test.exe --automated --gtest_filter=-ViERtpFuzzTest* --capture_test_ensure_resolution_alignment_in_capture_device=false

Review URL: https://webrtc-codereview.appspot.com/1025005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3345 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 09:13:00 +00:00
stefan@webrtc.org
fcd8585874 Enable external encoders with internal picture source.
CL enables registering of external encoder with internal picture
source on API by adding simple passthrough parameter that is already
supported within video engine.

Review URL: https://webrtc-codereview.appspot.com/1006006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-09 08:35:40 +00:00
mikhal@webrtc.org
658d423e81 Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers.
BUG=988

Review URL: https://webrtc-codereview.appspot.com/995014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3343 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-08 19:19:59 +00:00
elham@webrtc.org
27cb3017f5 Updated version number to 3.20
Review URL: https://webrtc-codereview.appspot.com/1023008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3341 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 21:54:41 +00:00
phoglund@webrtc.org
df3a15f63b Removed spaces from full stack test labels, consolidated graphs
NOTE TO SELF: save history on master when deploying!

BUG=
TEST=Ran locally

Review URL: https://webrtc-codereview.appspot.com/1021007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3337 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 10:06:52 +00:00
mflodman@webrtc.org
d73527ccab Changed assert to log.
Review URL: https://webrtc-codereview.appspot.com/1010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-20 09:26:17 +00:00
stefan@webrtc.org
1960219530 Make protection method, filename and resolution configurable for FullStackTest.
Review URL: https://webrtc-codereview.appspot.com/991007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3315 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-19 12:45:16 +00:00
mikhal@webrtc.org
0b18fb38e6 vie auto test: Adding a constructor for NetworkParameters
BUG=

Review URL: https://webrtc-codereview.appspot.com/995013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3310 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 19:47:52 +00:00
mikhal@webrtc.org
622c8bd0cc ViE autotest: Adding loss models to the external transport
Review URL: https://webrtc-codereview.appspot.com/1000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3309 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 17:21:51 +00:00
elham@webrtc.org
ad6845f4c4 Updated version number to 3.19
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/995007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3297 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 21:28:09 +00:00
mflodman@webrtc.org
4aee6b637d Added API to get receive side video delay.
BUG=1222

Review URL: https://webrtc-codereview.appspot.com/997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3294 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 14:02:10 +00:00
stefan@webrtc.org
32519398b6 Remove latency excl network and add render time diff stats.
Review URL: https://webrtc-codereview.appspot.com/996004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3290 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 09:03:27 +00:00
elham@webrtc.org
ddebc17bee Fix for buffer overflow, WebRTC issue 1196
Review URL: https://webrtc-codereview.appspot.com/998004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3286 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-13 21:55:47 +00:00
mflodman@webrtc.org
7acb65a870 Added jitter to fake network pipe.
Review URL: https://webrtc-codereview.appspot.com/988004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3283 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-13 15:53:11 +00:00
stefan@webrtc.org
91c91df35a Track the actual render time rather than the decode time.
Review URL: https://webrtc-codereview.appspot.com/993004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3282 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-13 15:26:01 +00:00
phoglund@webrtc.org
5b689efe8e Will now only require near-perfect PSNR and SSIM.
BUG=
TEST=Ran test and checked we accept somewhat lower values.

Committed: https://code.google.com/p/webrtc/source/detail?r=3269

Review URL: https://webrtc-codereview.appspot.com/964031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3278 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-13 10:15:06 +00:00
andrew@webrtc.org
d8aeb30d55 Revert 3269
> Will now only require near-perfect PSNR and SSIM.
> 
> BUG=
> TEST=Ran test and checked we accept somewhat lower values.
> 
> Review URL: https://webrtc-codereview.appspot.com/964031

TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3270 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 20:58:32 +00:00
phoglund@webrtc.org
735a6cec96 Will now only require near-perfect PSNR and SSIM.
BUG=
TEST=Ran test and checked we accept somewhat lower values.

Review URL: https://webrtc-codereview.appspot.com/964031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3269 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-12 15:20:34 +00:00
hclam@chromium.org
f222a00881 Use TRACE_EVENT to track time spent in VP8 encoding
Using the TRACE_EVENT macro to log VP8 encoding events.
Review URL: https://webrtc-codereview.appspot.com/968011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3264 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-11 22:27:55 +00:00
stefan@webrtc.org
71258c594b Add a third full stack test and support for random jitter in ext transport.
BUG=

Review URL: https://webrtc-codereview.appspot.com/975005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3260 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-11 15:14:56 +00:00
mflodman@webrtc.org
eaf7cf26fe Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced.
Review URL: https://webrtc-codereview.appspot.com/935032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3259 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-11 11:47:22 +00:00
leozwang@webrtc.org
8e49b02f3d Add more audio codec information into codec list
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/974009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3250 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-07 22:26:57 +00:00
fischman@webrtc.org
f4e070eca5 Added auto-call feature to WebRTCDemo.
This (compile-time switchable) option automatically starts & stops calls in
series to stress-test the setup/teardown codepaths.  When startCPULoad() is
removed (https://webrtc-codereview.appspot.com/972008/) this showed no
hangs/crashes after completing 200 start/stop pairs.

Also fixed a tiny shutdown-order bug (onDestroy() calling super.onDestroy()
before performing self-shutdown) and changed default video frame resolution to
640x480 to more effectively stress the device (and be a more compelling demo).

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/939032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3238 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 16:53:43 +00:00
stefan@webrtc.org
e861359925 Adds two full stack performance metrics for end-to-end delay.
Review URL: https://webrtc-codereview.appspot.com/937034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3235 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 09:40:46 +00:00
dwkang@webrtc.org
6bd737a714 First pass of MediaCodecDecoder which uses Android MediaCodec API.
Background:
As of now, MediaCodec API is the only public interface which enables us
to access low level HW resource in Android. ViEMediaCodecDecoder will be
used for further experiments/exploration.

TODO:
  To fix known issues. (detaching thread from VM and frequent GC)
Review URL: https://webrtc-codereview.appspot.com/933033

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 06:38:19 +00:00
fischman@webrtc.org
d814d71d92 Delete {start,stop}CPULoad() since they're broken.
- stopCPULoad is incorrect; since mIsBackgroudLoadRunning isn't declared
  volatile, the empty while loop in the background thread isn't required to do a
  memory read (as opposed to reading the value just once and caching it).  The
  result is that stopCPULoad() may never return as the .join() waits forever.
- startCPULoad isn't guaranteed to tax the CPU; the JVM is free to replace the
  while loop in startCPULoad() with a thread pause since it can prove it'll
  never exit the loop once entered (b/c of the previous item).

It's not clear what correct behavior here would be so I'm deleting the code
rather than trying to make it work.  This was responsible for at least most if
not all of the hanginess of start/stop'ing multiple calls in series.

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/972008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3202 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 23:00:41 +00:00
fischman@webrtc.org
be5b5ba490 Enable building WebRTCDemo apk using Release webrtc libs, take 2.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

Review URL: https://webrtc-codereview.appspot.com/966029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3201 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 18:06:00 +00:00
phoglund@webrtc.org
34dab50bb4 Corrected .h path.
TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/972009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3198 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 10:20:18 +00:00
phoglund@webrtc.org
273ccad59d Fixed standard PSNR/SSIM test.
BUG=1103

Review URL: https://webrtc-codereview.appspot.com/971005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3197 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 10:08:16 +00:00
stefan@webrtc.org
bf41508807 Properly remove the bitrate observer when ViEEncoder is destructed.
BUG=1090

Review URL: https://webrtc-codereview.appspot.com/969013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3196 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 09:18:53 +00:00
phoglund@webrtc.org
cde46fa5d2 Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this.
BUG=1152

Review URL: https://webrtc-codereview.appspot.com/964019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3194 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 08:22:45 +00:00
fischman@webrtc.org
de6f8fbd6d Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

TBR=leozwang@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/968010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3191 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 01:18:04 +00:00
fischman@webrtc.org
28afee04ae Enable building WebRTCDemo apk using Release webrtc libs.
Now passing BUILDTYPE=Release to both the make that builds the libs and the
ndk-build that builds the app makes the app use non-Debug libs.

Review URL: https://webrtc-codereview.appspot.com/972007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3190 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-29 01:09:44 +00:00
leozwang@webrtc.org
aa46ea0b8b Remove ringtone from test app
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/968009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3184 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-28 19:39:23 +00:00
kjellander@webrtc.org
52ec985d82 Fixing vie and voe auto test project paths for test execution.
By letting fileutils.h know the path to the executable, the tests will be able to find the project root dir and resource file paths even when the test is executed outside the checkout dir.

See http://review.webrtc.org/858014/ for more background.

Today, these tests are failing in the FYI waterfall since they are run "Chromium style" (i.e. from one level above the checkout dir). Since we're moving in that direction this needs to be fixed. It has been fixed for all other tests already.

TEST=Local test execution of vie_auto_test and voe_auto_test with CWD one level above trunk/

Review URL: https://webrtc-codereview.appspot.com/974004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3173 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-27 10:01:01 +00:00
elham@webrtc.org
6e46d5b1c1 Updated version number to 3.18
Review URL: https://webrtc-codereview.appspot.com/930027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3166 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 17:08:34 +00:00
mflodman@webrtc.org
7c894b7cc7 Wire up CallStats to provide modules with correct RTT.
BUG=769
TEST=Manual test since there is no ViE APi to get RTT for receive channels.

Review URL: https://webrtc-codereview.appspot.com/937027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 12:40:15 +00:00
stefan@webrtc.org
ad7f1fefad Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter.
BUG=

Review URL: https://webrtc-codereview.appspot.com/969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3157 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 11:19:29 +00:00
phoglund@webrtc.org
849fb8ebd2 Removed codec comparison test: it didn't work and probably never will.
The central problem is that we cannot sync the frames in the input video with the output video, which makes PSNR/SSIM go crazy. The test only appeared to succeed earlier due to a bug in the test. We can consider this a failed experiment, but we did learn a lot from it :)

BUG=550

Review URL: https://webrtc-codereview.appspot.com/969006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3155 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-23 09:21:44 +00:00