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Author SHA1 Message Date
kjellander@webrtc.org
38ebf98c2a Refactor barcode decoder to use Zxing's C++ version 
By using the C++ version of Zxing, we can avoid having Java and Ant
as a dependency when running Video quality analysis on the bots.
This makes it far more easy to setup automation on new machines.

I also moved the scripts into the webrtc/ folder so it will be synced by default when building in Chrome (eliminating the need of a separate solution).

This CL also removes the need of the FFMPEG_HOME variable and replaces
its use with a command line flag to make the tool run smoothly on
Windows.

BUG=none
TEST=locally running the script on Windows, Mac and Linux.

Review URL: https://webrtc-codereview.appspot.com/1099007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 10:58:21 +00:00
jansson@webrtc.org
755e19adfc - Checks the OS and runs the appropriate commands for Dummynet (ipfw) 
- Added pipe rule flush handling
- Also fixed a bug preventing any rule settings other than default from being 
  used no matter what preset was chosen
- Fixed some comments.

BUGS=none
TEST= Windows and linux
Review URL: https://webrtc-codereview.appspot.com/1158006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 10:50:14 +00:00
kjellander@webrtc.org
971278a962 Splitting out video_coding_test executable again. 
This CL undoes the merge of the developer test tool and the gtest tests
that was merged in https://code.google.com/p/webrtc/source/detail?r=3176

Doing that, we get a pure gtest executable of
video_coding_integrationtests which can run properly on the bots.

BUG=none
TEST=Trybots passing + local execution on Linux with:
out/Debug/video_coding_integrationtests --gtest_print_time (to ensure it will be possible to run with runtest.py)

Review URL: https://webrtc-codereview.appspot.com/1171007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-08 10:20:53 +00:00
wu@webrtc.org
3137a21068 Dtmf twinkle-twinkle. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1160005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 21:59:23 +00:00
andrew@webrtc.org
df123ed604 Roll libvpx 180104:186754. 
Picks up the ability to disable VP9 through gyp.

Review URL: https://webrtc-codereview.appspot.com/1162009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 21:27:41 +00:00
kjellander@webrtc.org
603a7f47a4 Add third_party/ dependencies to svn:ignore 
Adding the following directories to be ignored by SVN:
third_party/android_testrunner
third_party/android_tools
third_party/WebKit
This will speed up gclient sync/revert operations for the bots.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 20:31:45 +00:00
kma@webrtc.org
2951a6df4a Fixed an assembly code error in AECM for ARMv7. 
Possibly related to an AECM quality issue encountered at Chrome testing.
No bug was logged.
Review URL: https://webrtc-codereview.appspot.com/1160006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 18:25:34 +00:00
stefan@webrtc.org
84cd8e39cf Disable frame dropper for screenshare mode. 
BUG=1466

Review URL: https://webrtc-codereview.appspot.com/1170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:12:32 +00:00
stefan@webrtc.org
7c16c3c4a1 Move video_coding OWNERS to video_coding/. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 13:11:32 +00:00
phoglund@webrtc.org
5d37139374 Fixed a ton of Python lint errors, enabled python lint checking. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1166004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 09:59:43 +00:00
andrew@webrtc.org
52b57cc0d5 Fix debug file buffer bug introduced in r3574. 
This correctly uses int16_t rather than float. Only affects the debug
file buffer, not the production code path.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/1162008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-07 00:45:50 +00:00
mikhal@webrtc.org
efe4edb6da Enabling bufffering mode with no sync module or VoE 
BUG= 1454

Review URL: https://webrtc-codereview.appspot.com/1149006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 23:29:33 +00:00
braveyao@webrtc.org
488d4c9493 Submit symlink in apprtc from Linux since it fails from Win 
Review URL: https://webrtc-codereview.appspot.com/1169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 06:45:14 +00:00
braveyao@webrtc.org
07db4a6918 Add symlink of adapter.js from apprtc to base 
Review URL: https://webrtc-codereview.appspot.com/1160004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-06 03:35:03 +00:00
andrew@webrtc.org
a9a1df0035 Remove the error return on SetAGC failure introduced by r3605. 
BUG=webrtc:1464

Review URL: https://webrtc-codereview.appspot.com/1166005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 23:36:10 +00:00
fbarchard@google.com
64dc671167 Roll libyuv to r590 
BUG=none
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1161004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 21:02:33 +00:00
elham@webrtc.org
90eb5c84f9 1. Updated test pages to include Chrome Frame meta tag 
2. Updated test pages to use adapter.js
Review URL: https://webrtc-codereview.appspot.com/1142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 19:53:01 +00:00
bjornv@webrtc.org
91d11b3cdd Adds new AEC API to audio_processing. 
One unit test added.
Tested with audioproc_unittest and trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 16:53:09 +00:00
hta@webrtc.org
db3f42782c Using adapter.js and getRemoteStreams 
Needed to make the stats demo work on M26.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1165004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 15:23:40 +00:00
stefan@webrtc.org
1dc0aa2de2 Fix for build error on android introduced with r3609. 
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1164004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:30:47 +00:00
stefan@webrtc.org
a27107004d Split the NACK list into multiple RTCPs if it's too big. 
TEST=rtp_rtcp_unittests
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1148006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 09:02:06 +00:00
vikasmarwaha@webrtc.org
a856db26a6 Moved trace function to adapter.js and removed from pc1 & multiple.html. 
Review URL: https://webrtc-codereview.appspot.com/1156005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 03:35:26 +00:00
turaj@webrtc.org
24045c5a02 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. 
bug=issue1370
test=trybots
Review URL: https://webrtc-codereview.appspot.com/1121007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 03:14:22 +00:00
vikasmarwaha@webrtc.org
7881b574dd Updated path of adapter.js for dtmf & pc1-audio demos. 
TBR = wu@webrtc.org,juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1151005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 02:04:07 +00:00
andrew@webrtc.org
f0a90c37c4 Expose the capture-side AudioProcessing object and allow it to be injected. 
* Clean up the configuration code, including removing most of the weird defines.
* Add a unit test.

Review URL: https://webrtc-codereview.appspot.com/1152005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 01:12:49 +00:00
bjornv@webrtc.org
7f95732fe2 AEC Refactoring: Removes lint warning 
Changed inlude order.

TBR=andrew@webrtc.org
TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1156004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 23:47:39 +00:00
vikasmarwaha@webrtc.org
99f13464df Typo in index.html and updated svn propset for dtmf & pc1-audio demos. 
Review URL: https://webrtc-codereview.appspot.com/1145007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 19:34:46 +00:00
vikasmarwaha@webrtc.org
b203540e25 Redirect webrtc-demos.appspot.com to svn site and added dtmf & pc1-audio demos. Also updated index page to include information about new demos. 
Review URL: https://webrtc-codereview.appspot.com/1148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 18:57:09 +00:00
elham@webrtc.org
ec6226eedc Updated version number to 3.25 
Review URL: https://webrtc-codereview.appspot.com/1149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 18:05:56 +00:00
stefan@webrtc.org
a64300af50 Refactor NACK list creation to build the NACK list as packets arrive. 
Also fixes a timer bug related to NACKing in the RTP module which could cause packets to only be NACKed twice if there's frequent packet losses.

Note that I decided to remove any selective NACKing for now as I don't think the gain of doing it is big enough compared to the added complexity. The same reasoning for empty packets. None of them will be retransmitted by a smart sender since the sender would know that they aren't needed.

BUG=1420
TEST=video_coding_unittests, vie_auto_test, video_coding_integrationtests, trybots

Review URL: https://webrtc-codereview.appspot.com/1115006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 15:24:40 +00:00
phoglund@webrtc.org
17b867ae00 compile fix for get_nprocs() with uClibc 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1150006
Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 15:09:03 +00:00
phoglund@webrtc.org
44f85a49d8 Fixed coverity defects (CID 14657 and 14656). 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1153006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-04 14:59:31 +00:00
fischman@webrtc.org
73ec386d8a VideoCaptureAndroid can now capture just buffers without also rendering to a SurfaceView. 
This saves ~15% CPU on a Nexus 7 running AppRTCDemo.

BUG=1169

Review URL: https://webrtc-codereview.appspot.com/1150005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-03 17:28:03 +00:00
andrew@webrtc.org
2412085bc1 Don't upsample the capture signal early. 
* Remove the unneeded _mixingFrequency.
* Rename CheckForSendCodecChanges to better elucidate its function.
* Remove an unnecessary memcpy.

Upsampling should be done late in the chain. This is practically relevant
on mobile, where the capture rate is fixed at 16 kHz. When using Opus, the
signal was upsampled to 32 kHz and was no longer compatible with AECM, which only supports up to 16 kHz.

NEEDS_QA=true
TEST=run calls with a variety of capture device rates and codecs
BUG=chromium:178040,webrtc:1446

Review URL: https://webrtc-codereview.appspot.com/1146004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-02 00:14:46 +00:00
bemasc@google.com
ea386147f1 Update integration tests for idempotent RTP header settings. 
Review URL: https://webrtc-codereview.appspot.com/1152004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 23:43:14 +00:00
kma@webrtc.org
7d6f11302e Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools. 
Review URL: https://webrtc-codereview.appspot.com/1153004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 23:01:14 +00:00
andrew@webrtc.org
6be1e934ad Properly error check calls to AudioProcessing. 
Checks must be made with "!= 0", not "== -1". Additionally:
* Clean up the function calling into AudioProcessing.
* Remove the unused _noiseWarning.
* Make the other warnings bool.

BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 18:47:28 +00:00
leozwang@webrtc.org
9ee5a4ccd8 Enable External MediaProcessing on Mobile 
Bug=None
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1133005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 17:19:35 +00:00
bemasc@google.com
603ae3ece2 Make RtpHeaderExtensionMap::Register and ::Deregister idempotent. 
This CL changes the return code of these methods to indicate
success instead of failure when there is nothing to change.

This change appears to resolve an issue where enabling the
timestamp offset extension via SDP would result in a failure if
that extension had already been enabled.
Review URL: https://webrtc-codereview.appspot.com/1118008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 17:03:02 +00:00
andrew@webrtc.org
78693fe37c Return an error when greater than 16 kHz is used with AECM. 
BUG=chromium:178040

Review URL: https://webrtc-codereview.appspot.com/1146005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 16:36:19 +00:00
mflodman@webrtc.org
6648093911 Destroy VCM and VPM instead of delete. 
Review URL: https://webrtc-codereview.appspot.com/1149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 14:51:23 +00:00
phoglund@webrtc.org
527fb4d607 Revert "Will now run pylint on all python files if there's at least one modified python file in the checkin." 
This reverts commit 6bd8730dfad6e7c5a5cf9a089605fcb9f83a13e0.

TBR=ajm@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1150004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 14:08:28 +00:00
kma@webrtc.org
a0936a6e45 Limit ARM instruction "strheq" to Apple's clang compiler only. 
bug =
Review URL: https://webrtc-codereview.appspot.com/1111008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 00:20:10 +00:00
marpan@webrtc.org
7d052c3cb2 Turn off error concealment in videoprocessor_integration tests. 
Review URL: https://webrtc-codereview.appspot.com/1123006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 16:44:28 +00:00
braveyao@webrtc.org
6b6eb44cca Add supporting to V4L2_PIX_FMT_JPEG since it works same as MJPEG. 
ISSUE=529
TEST=unittest
Review URL: https://webrtc-codereview.appspot.com/1120006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 10:08:02 +00:00
stefan@webrtc.org
9e254133ad Rewrite the jitter buffer statistics test and put make it robust under valgrind. 
BUG=1158

Review URL: https://webrtc-codereview.appspot.com/1116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-28 08:45:23 +00:00
vikasmarwaha@webrtc.org
98fce15c6f Adding webrtc-sample demos under trunk/samples. 
Review URL: https://webrtc-codereview.appspot.com/1126005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 23:22:10 +00:00
bjornv@webrtc.org
132c15de30 AEC Refactoring: 
* Adds pointer to low level AecCore struct.
* Adds a simple unit test of this new call.

Tested with audioproc_unittest, trybots

TEST=none
BUG=none

Review URL: https://webrtc-codereview.appspot.com/1121006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 21:03:41 +00:00
stefan@webrtc.org
e1c4ed958d Fix to send a full NACK list at least roughly once every 1.5 x RTT. 
BUG=1434

Review URL: https://webrtc-codereview.appspot.com/1111007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 16:23:06 +00:00
kma@webrtc.org
83561fb173 Fixed a bug in WebRtcNsx_PrepareSpectrumNeon() for NS in ARM Neon platform. 
No written bug report.

Tested with Nexus-S. Issue disappeared with the change.
Review URL: https://webrtc-codereview.appspot.com/1126006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-27 01:16:44 +00:00