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Author SHA1 Message Date
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
leozwang@webrtc.org
be86a6d968 Explicitly disable sincos optimization on Android. 
I uploaded this CL before, now it turned out that although it's an
issue in compiler, but it will not be solved in short term, we have
to work around in our code termporally.

We can chat in person if you want to know more details.

BUG=
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1026006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-10 22:15:51 +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
kma@webrtc.org
0af0d3d3f4 Address a build issue with Android-Clang compiler: 
error: the value is truncated when put into register, use a modifier to specify the size [-Werror,-Wasm-operand-widths]
  __asm __volatile ("ssat %0, #16, %1" : "=r"(out16) : "r"(value32));
Review URL: https://webrtc-codereview.appspot.com/1029006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-10 00:46:37 +00:00
marpan@webrtc.org
ef1a760446 Rounding error fix in media_opt_util. 
Review URL: https://webrtc-codereview.appspot.com/1013006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 22:13:19 +00:00
andrew@webrtc.org
a5e7e76def Use %d for signed value in trace. 
BUG=1259
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1028007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 17:11:50 +00:00
andrew@webrtc.org
08d660f08e Allow for some error in volume testing. 
BUG=616
TESTED=voe_auto_test:VolumeTest.* now passes on a MacBook

Review URL: https://webrtc-codereview.appspot.com/1028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 17:07:02 +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
mflodman@webrtc.org
2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals. 
BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 13:54:43 +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
c38eef896a Reformatted RTPReceiver. 
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)

BUG=
TEST=Trybots, vie_ & voe_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/998007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-07 10:18:30 +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
stefan@webrtc.org
1ea4b502ef Refactor receiver.h/.cc. 
TEST=video_coding_unittests, vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/994008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-07 08:49:41 +00:00
andrew@webrtc.org
1926d33344 Change Sleep() comment in test fixture. 
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1023006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-05 03:30:11 +00:00
andrew@webrtc.org
bcb717428f .gitignore: Add *.mk, created as part of ChromiumOS build 
Contributed by Josh Triplett <josh.triplett@intel.com>

BUG=None
TEST=Build Chromium and ChromiumOS from source, and run "repo status",
     with and without this change.

Review URL: https://webrtc-codereview.appspot.com/1000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-04 21:25:42 +00:00
kma@webrtc.org
f545cf8f10 Addressing webrtc issue 1237, http://code.google.com/p/webrtc/issues/detail?id=1237. 
Code compared to C. Bit-exact.
Review URL: https://webrtc-codereview.appspot.com/1021004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-04 17:40:21 +00:00
phoglund@webrtc.org
6f62836ccf Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?) 
Revert "Further relax thresholds in mixing test."

This reverts commit 53c7e973a02d65e0b4981129e7ccfc145d955eda.

Revert "Fix implicit conversion error in mixing test."

This reverts commit 68d7e2258082d7d2b9461061e03e2f2d6ae78c4f.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 14:33:00 +00:00
phoglund@webrtc.org
5c8d9d30e2 Reformatted tick_util. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1014004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 09:50:17 +00:00
phoglund@webrtc.org
daabfd25a6 Reformatted trace* files. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1015004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 09:37:03 +00:00
andrew@webrtc.org
201d4b61d1 Fix implicit conversion error in mixing test. 
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 19:59:53 +00:00
andrew@webrtc.org
b2b628d5cd Further relax thresholds in mixing test. 
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 18:50:13 +00:00
andrew@webrtc.org
00c7c4315b Replace voice engine utility functions with system wrapper variants. 
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 16:06:39 +00:00
phoglund@webrtc.org
ec9c942e45 Reformatted thread and static_instance. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 08:45:03 +00:00
pwestin@webrtc.org
1b6da28047 Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests. 
Landing of 573005 On behalf of an1kumar@gmail.com

TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1002008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-21 17:46:24 +00:00
brykt@google.com
f556890844 Added possibility to repeat frames. Also added unittest for that feature. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/994005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 11:42:45 +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
tina.legrand@webrtc.org
d0d41498a3 Adding AUDIO application as default for Opus stereo 
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:23:10 +00:00
phoglund@webrtc.org
ad0ed582b5 Fixed a missed initialization (found by valgrind FYI bot). 
http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/327/steps/memory%20test%3A%20memcheck_voe_auto_test/logs/stdio

BUG=
TEST=Reproduced valgrind error, verified gone after fixing.

Review URL: https://webrtc-codereview.appspot.com/1008005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-20 09:14:36 +00:00
leozwang@webrtc.org
ac77084583 Roll opus to 172355 and delete opus_demo from webrtc opus 
opus_demo has been inlucded in opus in chromium.

BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/973013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 17:24:30 +00:00
phoglund@webrtc.org
6bc5d4dc07 Reformatted sort. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/998006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 14:55:24 +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
tina.legrand@webrtc.org
4275ab1ca0 Implement NetEq duration estimation for Opus. 
Review URL: https://webrtc-codereview.appspot.com/983004
Patch from Ralph Giles <giles@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 09:52:45 +00:00
leozwang@webrtc.org
515ef2428c Clean up variable after it gets deleted 
BUG=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/939038

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 05:36:36 +00:00
mikhal@webrtc.org
e239bf0940 Making I420VideoFrame ref-counted 
BUG=937
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 00:07:57 +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
phoglund@webrtc.org
6e0ce73741 Reformatted map classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 17:18:35 +00:00
phoglund@webrtc.org
61f39a3174 Fixed bad header name. 
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1001008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 16:02:13 +00:00
phoglund@webrtc.org
07bf43c673 Replaced the _audio parameter with a strategy. 
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 15:40:53 +00:00
phoglund@webrtc.org
59ad541e57 Reformatted rw_lock classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1007004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 15:20:35 +00:00
kjellander@webrtc.org
10abe25f6d Make audioproc output files be written to output dir by default. 
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat

TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1003005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-17 18:28:07 +00:00
fbarchard@google.com
3c37354b70 Initialize 3 variables which are preventing VS2012 from building. 
BUG=1211
TESTED=ninja -C out\Release
Review URL: https://webrtc-codereview.appspot.com/992005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-15 01:09:18 +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
hclam@chromium.org
c5fcb0879b Update trace_event.h to match the one in Chromium 
Chromium's trace_event.h has updated to remove some not-well-used features.
Update WebRTC's copy to match.
Review URL: https://webrtc-codereview.appspot.com/995006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 21:16:46 +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
phoglund@webrtc.org
1c75918302 Disabled flaky test. 
From flake in http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/270

Review URL: https://webrtc-codereview.appspot.com/1001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 10:40:05 +00:00
phoglund@webrtc.org
7659d914bb Decoupled video rtp receiver from rtp receiver. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/995005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:57:37 +00:00