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Author SHA1 Message Date
bjornv@webrtc.org
8526459a2e Added tests for multiple near-end support. 
TEST=trybots, audioproc_unittest
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1063007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 22:33:17 +00:00
bjornv@webrtc.org
57f3a11958 Short CL: only name change. 
From |handle| to |self| for consistency.

BUG=None

Review URL: https://webrtc-codereview.appspot.com/1072005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 22:02:15 +00:00
bjornv@webrtc.org
94c213af1a Separated far-end handling in BinaryDelayEstimator. 
This CL is one step in a larger change of the DelayEstimator where we will open up for multiple near-end signals.

This particular CL separates the low level far-end parts without affecting the usage externally. This is a first step towards separating the far-end and near-end parts giving the user the control.

BUG=None
TEST=audioproc_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1068005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 15:53:41 +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
kjellander@webrtc.org
d3ecb615ba Fix path to perf Python scripts in test.gyp 
The path in test.gyp in r3411 was incorrect since it was based on the symlink that does not exist on Windows.
This CL changes it to reference the actual path in /tools/perf

TBR=phoglund
BUG=none
TEST=win try bot.

Review URL: https://webrtc-codereview.appspot.com/1074006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 12:18:15 +00:00
phoglund@webrtc.org
43da54a458 Reformatted rtp_sender: made lint clean. 
TESTED=rtp_rtcp_unittests
BUG=

Review URL: https://webrtc-codereview.appspot.com/1062004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 10:53:38 +00:00
kjellander@webrtc.org
3e47a0a611 Test launching script 
This script is an attempt to move flags and argumetns to tests from the
buildbot configuration to the source tree.
This will make it easier for anyone to modify test execution behavior
and also has the benefit that it's easier to run the tests in a similar
fashion on a developer workstation.

NOTICE: The audio comparison tool will need to be moved to ~/bin when bots are going to switch over to using this script for execution.

TEST=local execution.
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1021006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 10:10:53 +00:00
kma@webrtc.org
c4373bc737 Moved several function pointer declarations in iSAC to isac initialization file. 
Fixed clang linker problem of not being able to find symbols.
Review URL: https://webrtc-codereview.appspot.com/1061006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 04:55:21 +00:00
kma@webrtc.org
16d540eff1 Fixed text relocation code related to ARM assembly code. 
Refer to WebRTC issue 1300.
Review URL: https://webrtc-codereview.appspot.com/1055004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 03:18:05 +00:00
kma@webrtc.org
e8482f0e9f Revert 3406 
> Moved all function pointer declarations in iSAC to a single place.
> Review URL: https://webrtc-codereview.appspot.com/1057006

TBR=kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 23:57:56 +00:00
niklas.enbom@webrtc.org
cd2f1356ee Revert 3405 
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 22:05:30 +00:00
kma@webrtc.org
ebef7e4ac1 Moved all function pointer declarations in iSAC to a single place. 
Review URL: https://webrtc-codereview.appspot.com/1057006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 21:19:24 +00:00
niklas.enbom@webrtc.org
05e7bfeeea Mainly hlundin's patch. 
Review URL: https://webrtc-codereview.appspot.com/1052004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 18:53:43 +00:00
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
phoglund@webrtc.org
8382ad557b Added perf expectations for stack tests. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1043006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-22 12:19:24 +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
phoglund@webrtc.org
899699e6f3 Enabled full lint checking for ALL WebRTC changes. 
According to decision at the 14/1 -13 test sync meeting.

TESTED=Made local modification; noted the brutal amount of presubmit lint warnings.
BUG=

Review URL: https://webrtc-codereview.appspot.com/1063004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-21 15:57:34 +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
phoglund@webrtc.org
4ad64458cb Fixed URL unquoting in bot names. Added iOS Device. Removed unnecessary filter code. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1046005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 13:44:21 +00:00
kjellander@webrtc.org
c39962aa8d Adding TRYSERVER_ROOT to codereview.settings 
This is needed for tryjobs to work with updated trybot configurations.

BUG=webrtc:1309
TEST=Submitted try jobs and verified the patch applies properly.

Review URL: https://webrtc-codereview.appspot.com/1045004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 08:27:48 +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
fbarchard@google.com
ebc6d8f172 libyuv r540 roll for valgrind tools update, optimized ARGBToI444_SSSE3 and I420Copy single memcpy per plane if contiguous. 
BUG=none
TEST=try bots still pass
Review URL: https://webrtc-codereview.appspot.com/1019012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 01:43:38 +00:00
hclam@chromium.org
00c18dbcca Fix libvpx for Android 
Android writes .a files to a different directory so update it accordingly.

BUG=1294
TEST=Builds on Android
Review URL: https://webrtc-codereview.appspot.com/1013013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-16 22:56:08 +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
kjellander@webrtc.org
14d1898bf9 Removing arena_thread_freeres suppression 
It is no longer needed since we are now using a Chromium revision that
is newer than
http://src.chromium.org/viewvc/chrome?view=rev&revision=172313
In that revision, the arena_thread_freeres suppression was added to
ignore.txt.

BUG=300
TEST=tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -t
out/Release/system_wrappers_unittests
and trybot execution on linux_tsan

Review URL: https://webrtc-codereview.appspot.com/1019010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-16 10:49:31 +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
kjellander@webrtc.org
75ba51938c Updating chromium_revision 169394:176094 
This resolves ninja error on Windows due to recent depot_tools update, since a newer GYP will be synced.

Initially we ran into a compile issue with libvpx: http://code.google.com/p/webm/issues/detail?id=521
The change in libvpx.gyp is needed since the newer version of Clang that
is used with this Chromium revision provides the -fsanitize=address flag
instead of the old (now deprecated) -faddress-sanitizer. Without
disabling them when asan=1, we'll get compile errors for the assembly
offsets generated for Libvpx. See http://crbug.com/159580 for more details.

BUG=libyuv:173
TEST=trybots passing.

Review URL: https://webrtc-codereview.appspot.com/1026004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-11 16:04:50 +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