Commit Graph

682 Commits

Author SHA1 Message Date
pwestin@webrtc.org
5621057956 Removing unused code.
Review URL: https://webrtc-codereview.appspot.com/349008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1442 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 12:45:47 +00:00
pwestin@webrtc.org
df9bd9bdbd Removed dead code.
Review URL: https://webrtc-codereview.appspot.com/352010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1437 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 11:42:02 +00:00
pwestin@webrtc.org
aafa5a331c Coverty report: Unititialized members
Review URL: http://webrtc-codereview.appspot.com/349007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1436 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 07:07:37 +00:00
asapersson@webrtc.org
43b8fc5c0d Review URL: http://webrtc-codereview.appspot.com/345011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1435 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 13:49:04 +00:00
stefan@webrtc.org
8ddf9a4e18 Ported more jitter buffer tests to unit tests.
BUG=
TEST=jitter_buffer_unittest

Review URL: http://webrtc-codereview.appspot.com/350009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1433 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 11:59:01 +00:00
asapersson@webrtc.org
869ce2d441 Review URL: http://webrtc-codereview.appspot.com/353002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1432 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 11:58:36 +00:00
asapersson@webrtc.org
0b3c35a258 Review URL: http://webrtc-codereview.appspot.com/321011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1431 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 11:06:31 +00:00
henrika@webrtc.org
f75901fa4c Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW).
You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 08:45:42 +00:00
andrew@webrtc.org
c8d012fb32 Use -msse2 for SSE2 optimized code.
When targeting 32-bit Linux, we need to pass -msse2 to gcc to compile
SSE2 intrinsics. However, -msse2 also gives gcc license to automatically
generate SSE2 instructions wherever it pleases. This will crash our code
on processors without SSE2 support.

This change breaks the files with SSE2 intrinsics into separate targets,
such that we can limit the scope of -msse2 to where it's needed.

We no longer need to employ the WEBRTC_USE_SSE2 define; the build system
decides when SSE2 is supported and compiles the appropriate files.

TBR=bjornv@webrtc.org
TEST=audioproc (performance testing), audioproc_unittest, video_processing_unittests, build on Linux (targeting ia32/x64, with disable_sse2==0/1), Mac, Windows

Review URL: http://webrtc-codereview.appspot.com/352008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1425 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 19:43:09 +00:00
andrew@webrtc.org
ee3fe5b982 Remove unused variable from mixer module.
R=henrike@webrtc.org
BUG=coverity-10288

Review URL: http://webrtc-codereview.appspot.com/344010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1424 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 17:54:57 +00:00
braveyao@webrtc.org
5f9a7baaea Review URL: http://webrtc-codereview.appspot.com/347012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1423 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 10:22:44 +00:00
mflodman@webrtc.org
117c119501 Only update REMB value if there is a calid bitrate estimate.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/352005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1421 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 08:52:58 +00:00
henrik.lundin@webrtc.org
33d5f69d5e Fix issue 218 with new solution
This one is slightly more elegant and efficient.

BUG=http://code.google.com/p/webrtc/issues/detail?id=218
TEST=

Review URL: http://webrtc-codereview.appspot.com/344009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1420 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 07:46:50 +00:00
andrew@webrtc.org
7859e10985 Propagate decoding errors to the mixer module.
Review URL: http://webrtc-codereview.appspot.com/348001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1417 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-13 00:30:11 +00:00
stefan@webrtc.org
c8277db7c8 Fix selective retransmissions after corrupt merge in r1373.
BUG=228
TEST=

Review URL: http://webrtc-codereview.appspot.com/345006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1414 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 15:38:50 +00:00
pwestin@webrtc.org
9cbe6867e7 Removed experiment.
Review URL: http://webrtc-codereview.appspot.com/345005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1413 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 15:35:28 +00:00
stefan@webrtc.org
ad4af57abd Fixes a jitter buffer NACK bug.
If no frame has been decoded the jitter buffer might generate huge
erroneous NACK lists.

Adds a couple of new jitter buffer unittests (some ported from
jitter_buffer_test.cc).

Adds a test to the VCM robustness tests.

BUG=226
TEST=VCMRobustnessTest, TestJitterBufferFull, TestNackListFull, TestNackBeforeDecode, TestNormalOperation

Review URL: http://webrtc-codereview.appspot.com/352002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1412 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 15:16:49 +00:00
mflodman@webrtc.org
80d60420ff RTCPSender::_bitrate_observer not initialized.
BUG=227
TEST=Valgrind

Review URL: http://webrtc-codereview.appspot.com/352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1410 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 14:28:53 +00:00
henrik.lundin@webrtc.org
053c7991e3 Add minimum waiting time to NetEQ metrics
Adding minWaitingTimeMs to ACMNetworkStatistics and to
NetworkStatistics. Also adding unittest.

TEST=audio_coding_unittests

Review URL: http://webrtc-codereview.appspot.com/350006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1408 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 14:16:44 +00:00
kjellander@webrtc.org
b39a3b4a7a Restoring unintentially renamed MS DirectShow source files in
http://webrtc-codereview.appspot.com/348005/

BUG=
TEST=Compiling on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/352003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1406 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 12:22:03 +00:00
kjellander@webrtc.org
7f3c724e12 Renaming 47 files from .cpp to .cc
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future.

BUG=
TEST=Compiling on Linux.

Review URL: http://webrtc-codereview.appspot.com/348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 10:23:41 +00:00
kjellander@webrtc.org
93546f8ed6 Removing unused file
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/347010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1404 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 10:00:33 +00:00
niklas.enbom@webrtc.org
553657b2f8 See http://codereview.chromium.org/9188022/ for details
Review URL: http://webrtc-codereview.appspot.com/347009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1403 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 08:49:34 +00:00
andrew@webrtc.org
83c7f6db0e Add missing file to iSAC gyp.
TBR=kma@webrtc.org
TEST=Linux build

Review URL: http://webrtc-codereview.appspot.com/344008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1394 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 20:16:32 +00:00
andrew@webrtc.org
921321ff62 Fix unused-variable warning in iSAC.
TBR=kma@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/348006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1393 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 19:50:20 +00:00
kma@webrtc.org
badf2b8044 Optimized an AR function in iSAC fix for ARMv7 (not Neon) platforms.
Bit exact. Speed doubled.
Review URL: http://webrtc-codereview.appspot.com/327001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1392 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 18:01:39 +00:00
mflodman@webrtc.org
04c18cb37a Update all child modules of with received bandwidth estimate.
BUG=224

Review URL: http://webrtc-codereview.appspot.com/347007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1391 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 14:18:33 +00:00
stefan@webrtc.org
cd8cea50a6 Fix decode error in NACK/FEC mode after network glitch.
Caused when recyclingframes until the next key frame to
regain frame buffers when the jitter buffer is full.

BUG=http://code.google.com/p/webrtc/issues/detail?id=225
TEST=

Review URL: http://webrtc-codereview.appspot.com/350005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1390 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 14:17:44 +00:00
perkj@webrtc.org
ce5990cb0b Fix defect http://code.google.com/p/webrtc/issues/detail?id=222
"ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets.

BUG=222
TEST= tested in loopback. No new test added yet.

Review URL: http://webrtc-codereview.appspot.com/343003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 13:00:08 +00:00
tina.legrand@webrtc.org
6b6ff558a8 Implementation if mono-to-stereo and vice versa in ACM.
Added stereo-to-mono and mono-to-stereo tests to end of TestStereo.cpp.

BUG=Aim to resolve issue 207, "Investigate stereo capture handling in modules"
TEST=audio_coding_module_test

Review URL: http://webrtc-codereview.appspot.com/345002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1385 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 10:12:54 +00:00
braveyao@webrtc.org
e3eaf44ccf one logical enhancement in CoreAudio error handler. It should never happen, but so far the only suspect to a rare crash report.
Review URL: http://webrtc-codereview.appspot.com/349002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1382 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 03:07:52 +00:00
stefan@webrtc.org
c5b73e3974 Further cleanup of OverUseDetector. Removed member no longer used.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1377 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 16:42:09 +00:00
pwestin@webrtc.org
a1783598a7 Bugfix for clang.
Review URL: http://webrtc-codereview.appspot.com/351001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1375 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 15:33:40 +00:00
pwestin@webrtc.org
5d35ceb34a Bugfix array length in test.
Review URL: http://webrtc-codereview.appspot.com/343007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1374 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 15:06:09 +00:00
pwestin@webrtc.org
8281e7dd4a Added RTX to ViE.
Review URL: http://webrtc-codereview.appspot.com/336001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1373 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 14:09:18 +00:00
tina.legrand@webrtc.org
ac4eb046e3 Added registration of RED and CNG to NetEq slave.
Bug found when working on issue 221. Missing registration of CNG was the cause of the bad audio (master and slave out of sync) reported in the issue.

NOTE! File has not been refactored to follow Google style.

BUG=http://code.google.com/p/webrtc/issues/detail?id=221

Review URL: http://webrtc-codereview.appspot.com/342006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1372 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 13:59:55 +00:00
henrik.lundin@webrtc.org
d4e8c0b3ff Fixing Issue 218
Taking care of the case when the raw waiting times vector from
NetEQ is zero length.

Also adding a new unittest to cover this case.

BUG=http://code.google.com/p/webrtc/issues/detail?id=218
TEST=AcmNetEqTest.TestZeroLengthWaitingTimesVector

Review URL: http://webrtc-codereview.appspot.com/349003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1370 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 13:46:06 +00:00
asapersson@webrtc.org
c5a1cee73e Review URL: http://webrtc-codereview.appspot.com/348004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1367 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 12:54:44 +00:00
stefan@webrtc.org
727e1611ac Removes debug file writing.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/343006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1365 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 11:45:47 +00:00
stefan@webrtc.org
b07aa403b3 Fixes issue 210. Removes diff between two different arrays.
Also fixes the FrameBuffer copy constructor.

BUG=210
TEST=

Review URL: http://webrtc-codereview.appspot.com/347002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1364 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 11:45:05 +00:00
stefan@webrtc.org
e21a8cf4d4 Fix issue 211: Make sure we always generate at least one FEC packet per frame if we need protection.
BUG=211
TEST=

Review URL: http://webrtc-codereview.appspot.com/348002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1363 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 08:37:33 +00:00
marpan@webrtc.org
2dad3fbe49 Media-opt: Added a filter type mode for the filtering of the received packet loss. This makes the filter selection explicit and easier to modify/test.
Removed the function UpdateLossPr(); the filter updates are done in the same function that returns the filtered loss.
Review URL: http://webrtc-codereview.appspot.com/333018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1361 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 18:18:36 +00:00
pwestin@webrtc.org
12d97f6637 Made send pad data generic (audio and video)
Review URL: http://webrtc-codereview.appspot.com/343001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1346 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-05 10:54:44 +00:00
pwestin@webrtc.org
3aa25de346 Bugfix OnNetworkChanged not triggered for RTCP compund messages if TMMBR is higher than last value.
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/342001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1344 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-05 08:40:56 +00:00
wu@webrtc.org
d6b827a28e Fix for the build broken on Windows.
Review URL: http://webrtc-codereview.appspot.com/335017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1341 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 22:38:05 +00:00
mikhal@webrtc.org
a58888d382 Updating capture module following latest libyuv api changes
Review URL: http://webrtc-codereview.appspot.com/337009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1338 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 19:23:24 +00:00
mikhal@webrtc.org
7d5ca2be1f Updating render module following latest libyuv api changes.
Review URL: http://webrtc-codereview.appspot.com/331019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1337 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 19:01:48 +00:00
kma@webrtc.org
746f9e31c0 Changed build settings for ARMv5 in Android.
I found some issues in building ARMv5 with ICM. This CL includes fixes,
and a design change which now will exclude any NEON libraries unless 
the build is for dynamic detection or for Neon specifically.
Review URL: http://webrtc-codereview.appspot.com/330021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1335 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 17:47:57 +00:00
pwestin@webrtc.org
6c1d41583a Fix for RTP extension audio level.
Review URL: http://webrtc-codereview.appspot.com/339002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 17:04:51 +00:00
andrew@webrtc.org
d77a6614fa Consts can't be used as C array size initializers.
(Unless you happen to be using clang...)

TBR=bjornv@webrtc.org
TEST=build on gcc

Review URL: http://webrtc-codereview.appspot.com/333029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1333 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 16:22:24 +00:00