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Author SHA1 Message Date
kma@webrtc.org
4782911572 Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. 
Review URL: https://webrtc-codereview.appspot.com/1005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 01:37:33 +00:00
henrik.lundin@webrtc.org
5dfb1f2cd3 Bug fix in WebRtcOpus_DurationEst 
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.

BUG=1334
TEST=try bots

Review URL: https://webrtc-codereview.appspot.com/1069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-23 11:57:03 +00:00
kjellander@webrtc.org
8126602e26 Fix frame_editing_unittest.cc 
The test fails since it's assuming out/testfile.yuv exists when running the test. Just opening the file at a later time than the SetUp function seems to break the test so that's not a viable solution. This CL uses a simple workaround that simply truncates the file before opening it, which works.

BUG=none
TEST=tools_unittests in Debug+Release on Mac, Win and Linux + memcheck, tsan, asan.

Review URL: https://webrtc-codereview.appspot.com/1067004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 22:45:59 +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
henrike@webrtc.org
09738616de Fixes payload spelling error. 
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/1052006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 16:43:45 +00:00
phoglund@webrtc.org
5accd370e7 RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. 
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/1058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 12:31:01 +00:00
andrew@webrtc.org
ae1a58bba4 Replace AudioFrame's operator= with CopyFrom(). 
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders.

Review URL: https://webrtc-codereview.appspot.com/1031007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 04:44:30 +00:00
stefan@webrtc.org
a678a3baee Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. 
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-21 07:42:11 +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
bjornv@webrtc.org
bb599b7089 This CL includes part of changes in a larger one. The final goal is to allow multiple delay estimators using the same reference (far-end) to save computational complexity. 
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1024010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 23:16:46 +00:00
bjornv@webrtc.org
a2d8b75f29 An API to get the internal estimation quality in the delay estimator has been added. Unit tests have been updated. There is no impact to other parts in WebRTC. 
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1036004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 21:54:15 +00:00
wjia@webrtc.org
2e2a4cff18 Remove <(library) from gyp file. 
This is a corresponding change from Chome.
Review URL: https://webrtc-codereview.appspot.com/1053004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 17:13:47 +00:00
henrike@webrtc.org
a3e6bec23a Posix Thread: Removes the setting of the run function to NULL which could cause data race. 
BUG=http://code.google.com/p/chromium/issues/detail?id=103711
TESTED=Code analysis (no tools)

Review URL: https://webrtc-codereview.appspot.com/1008006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 16:39:21 +00:00
niklas.enbom@webrtc.org
218c542c0b Make VoE handle longer delays 
Review URL: https://webrtc-codereview.appspot.com/1047004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 22:25:49 +00:00
mflodman@webrtc.org
c7e935f5eb Adding timeEndPeriod to Synchronize function, see bug for details. 
BUG=748
TEST=Win try bots.

Review URL: https://webrtc-codereview.appspot.com/1043005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 17:12:50 +00:00
phoglund@webrtc.org
efae5d5901 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead. 
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.

BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1022011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 16:10:45 +00:00
stefan@webrtc.org
20ed36dada Break out RtpClock to system_wrappers and make it more generic. 
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 14:01:20 +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
stefan@webrtc.org
a4b58860b7 Add a counter to the video rtp play output filename. 
Review URL: https://webrtc-codereview.appspot.com/1040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 09:27:17 +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
phoglund@webrtc.org
acfdd96ee3 Reformatted rtp_rtcp_impl*. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1035004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-16 10:27:33 +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
phoglund@webrtc.org
a22a9bd9ca Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. 
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-14 10:01:55 +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
wjia@webrtc.org
b119369cdc Fix android clang build. 
no-builtin-cos|sin|cosf|sinf are not used for some files (g711.c, g711_interface.c, g722_encode.c, g722_decode.c, g722_interface.c, pcm16b.c).
Review URL: https://webrtc-codereview.appspot.com/1032006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-12 01:52:07 +00:00
wjia@webrtc.org
3f9db3735e Fix android clang build. 
Android clang build complains about unused private field.
Review URL: https://webrtc-codereview.appspot.com/1025006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-12 01:09:03 +00:00
andrew@webrtc.org
bafdae3cfc Fix simulated analog gain in audioproc. 
* It doesn't make much sense to apply at all when reading from the protobuf.
* Reduced the gain to be closer to actual mics.

BUG=1260

Review URL: https://webrtc-codereview.appspot.com/1027007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 23:11:29 +00:00
andrew@webrtc.org
f908011eb4 Remove extra line. 
TBR=elham

Review URL: https://webrtc-codereview.appspot.com/1024008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 22:39:55 +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
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