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Author SHA1 Message Date
stefan@webrtc.org
03b2b85b19 Enable multi-res encoding feature in libvpx. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/761005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-09-03 06:57:01 +00:00
andrew@webrtc.org
d84d11160e Add WEBRTC_POSIX and remove unneeded defines. 
Review URL: https://webrtc-codereview.appspot.com/768007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 23:17:53 +00:00
marpan@webrtc.org
1e68577499 Update the 3 layer temporal pattern and apply the current fix to issue 6992310. 
Review URL: https://webrtc-codereview.appspot.com/753006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 21:17:12 +00:00
phoglund@webrtc.org
31eac6082d Disabled one more flaky test on Linux. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/770006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 13:49:23 +00:00
vspasova@webrtc.org
992da29c75 A tool that crops the top pixel rows of a YUV video. 
TEST=./tools/barcode_tools/yuv_cropper.py --width=640
--height=480 --crop_height=32
--yuv_file=<path_and_name_to_yuv_file>
--output_file=<path_and_name_to_output_file>

Review URL: https://webrtc-codereview.appspot.com/765005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 09:47:47 +00:00
kjellander@webrtc.org
6307dbf468 Updates the PRESUBMIT.py to allow years from 2011 to the current year. 
Since Chromium has moved to this policy, we should too.
Code is copied from /depot_tools/presubmit_canned_checks.py but modified for our purpose.

BUG=
TEST=Tested git cl presubmit with a modified .cc file with the 2011 header and one with the 2012.

Review URL: https://webrtc-codereview.appspot.com/770005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 07:07:11 +00:00
fbarchard@google.com
3baa8a2102 Roll libyuv to r338 which includes more complete unittests, lint fixes, and a bug fix for YUY2ToARGB which was oversampling by 1 row for UV channels. This should help with setting up build bots for libyuv and keeps the version in sync with chromium. 
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/771005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 04:12:08 +00:00
wu@webrtc.org
8a9f0f4e4d * Update to the peerconnection client to use jsep01. (Chromium 153489.) 
* Remove the peerconnection_server target from peerconnection.gyp since we have it in libjingle.gyp.
* Add enabled_libjingle_device_manager in supplement.gypi to add devicemanger to stand alone build.
* Add link settings to base.gyp which is needed by the new changes in peerconnection_client.

Note: Resolving hostname function has some problem on Windows in this revision.
So with this revision the peerconnection client can only take ip address directly as
the server address on Windows.
Review URL: https://webrtc-codereview.appspot.com/753008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-31 00:17:53 +00:00
vspasova@webrtc.org
1b0a02e12e Fixing a path and Ant invocation issue in build_zxing.py and delete_file issue in helper_functions.py 
Review URL: https://webrtc-codereview.appspot.com/761006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-30 12:56:38 +00:00
andrew@webrtc.org
16cfbe2e89 Fix a potential Chrome crash in audioproc. 
There are Chrome crashes which appear to be occurring during some kind
of teardown. We might be able to avoid them by locking in the destructor.

On the other hand, this might have no impact, but at least isn't a bad
thing to do.

BUG=chromium:145341

Review URL: https://webrtc-codereview.appspot.com/768005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 16:58:25 +00:00
phoglund@webrtc.org
4436fcebaf Disabled audio device tests on mac so we can launch the test on the bots. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/762004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 09:36:50 +00:00
andrew@webrtc.org
cc53b7c97b Disable test causing race conditions. 
TBR=kjellander
BUG=issue788

Review URL: https://webrtc-codereview.appspot.com/770004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 06:55:21 +00:00
andrew@webrtc.org
b93522857c Trivial fix for memcheck error. 
TBR=xians

Review URL: https://webrtc-codereview.appspot.com/763005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 04:23:14 +00:00
andrew@webrtc.org
55c0d4a683 Add support for clock drift compensation. 
Support clock drift compensation on Windows and add an API to allow
enabling dynamically.

BUG=issue773
TEST=unittest, trybots

Review URL: https://webrtc-codereview.appspot.com/744007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-29 02:13:12 +00:00
mikhal@webrtc.org
954cf806d9 Adding the video debug api to vie test record 
Review URL: https://webrtc-codereview.appspot.com/763004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 20:55:10 +00:00
andrew@webrtc.org
12dbcd9798 Roll Chromium 143348:152335. 
Fix libvpx-Android build error.

Review URL: https://webrtc-codereview.appspot.com/760004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 20:34:57 +00:00
fbarchard@google.com
a4fa46108c Bump libyuv version to 331 which includes clang build fixes. 
BUG=none
TEST=devenv -build Debug webrtc.sln
Review URL: https://webrtc-codereview.appspot.com/766004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 18:29:03 +00:00
mikhal@webrtc.org
e41bbdfecc Adding an API that allows recording of video data 
removing vie_codec from cl

Moving debug call from Codec to File impl.

Updating cl following review

Updating file name

Updating cl following review.

Updating CL following review.

Adding an API that allows recording of video data

updating cl

Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 16:15:16 +00:00
andrew@webrtc.org
7b2e919f7d Add pinned dep for cygwin. 
Review URL: https://webrtc-codereview.appspot.com/760005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 15:56:40 +00:00
stefan@webrtc.org
715faaf5ce Change so that capture_time_ms == 0 is considered invalid. 
No transmission offset computed if capture_time_ms is less than or equal to zero.

BUG=

Review URL: https://webrtc-codereview.appspot.com/761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 15:20:39 +00:00
kma@webrtc.org
94771cb6a0 Change preprocessor define WEBRTC_ANDROID into WEBRTC_ARCH_ARM, for ARM platform depended code. 
Review URL: https://webrtc-codereview.appspot.com/735010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 04:09:50 +00:00
kma@webrtc.org
7611791ade Added unit tests for several SPL macros/functions, and detailed all factors 
contributing to bit-not-exact between ARM assembly and generic C versions
in iSAC and SPL, by code comments.
Review URL: https://webrtc-codereview.appspot.com/741004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 00:43:55 +00:00
andrew@webrtc.org
2bea04076c Fix ia32 sse2 error in libvpx. 
denoising_sse2.c needs to be built with -msse2. We accidentally left it
in libvpx_src_x86.gypi in the last libvpx update.

TBR=stefan

Review URL: https://webrtc-codereview.appspot.com/751007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-28 00:05:34 +00:00
kma@webrtc.org
afdcc2c845 Added unit test for a filter bank function in iSAC-Fix, with some optimization 
in the assembly code.
Review URL: https://webrtc-codereview.appspot.com/754004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 23:54:13 +00:00
mikhal@webrtc.org
36b95b4753 Adding a recording tool to vie autottest 
Review URL: https://webrtc-codereview.appspot.com/746006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 21:39:50 +00:00
andrew@webrtc.org
36d0baf208 Fixes in preparation for next Chromium roll. 
Review URL: https://webrtc-codereview.appspot.com/743004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 20:58:56 +00:00
phoglund@webrtc.org
6b84b613d5 Disabling flaky tests in preparation of launching test on bots. 
BUG=777

Review URL: https://webrtc-codereview.appspot.com/745005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 15:16:20 +00:00
vspasova@webrtc.org
ac410e26a5 A small tool to run PSNR and SSIM between two I420 videos. 
The tool runs the analysis between every two corresponding frames until either of the videos runs out of frames.

The results are written in a file (results_file provided from the command line) in the format:
Frame: <frame_number>, PSNR: <psnr_value>, SSIM: <ssim_value>

TEST=
psnr_ssim_analyzer --reference_file=<name_of_file>
--test_file=<name_of_file> --results_file=<name_of_file>
--width=<width_of_frames> --height=<height_of_frames>

Review URL: https://webrtc-codereview.appspot.com/748007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 14:57:19 +00:00
hta@webrtc.org
4368c26b1a Changing to use SleepMs throughout audio_device module. 
Tested: Build audio_device_test_api and audio_device_test_func
Ran audio_device_test_api. Executed a random set of functions in
audio_device_test_func.

BUG=603

Review URL: https://webrtc-codereview.appspot.com/756008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-27 08:33:33 +00:00
henrike@webrtc.org
f7884f9900 Revert 2660 - updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/752007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-25 02:00:19 +00:00
henrike@webrtc.org
3387b88595 Makes it possible to disable frame dropping in the VP8 codec. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/757006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-24 23:01:38 +00:00
mikhal@webrtc.org
6a6121c0b1 updating cl 
Adding debug options

BUG=

Review URL: https://webrtc-codereview.appspot.com/751005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-24 22:08:25 +00:00
elham@webrtc.org
2a74de1e78 Bump version number to 3.11 
Review URL: https://webrtc-codereview.appspot.com/744005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-23 16:50:57 +00:00
stefan@webrtc.org
727a9f5648 Roll libvpx to c522217d. 
Also changing generate_gypi.sh to sort the list of files before writing gypis.

TBR=andrew@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/751004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 20:14:33 +00:00
stefan@webrtc.org
1ec2b38926 VP8 config tunings for improved quality. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/750004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 11:09:07 +00:00
henrika@webrtc.org
8a2fc88459 Adds new GetRemoteRTCPSenderInfo() and GetRemoteRTCPReportBlocks APIs to VoE. 
BUG=559
TEST=manual tests using Windows UI client.

Review URL: https://webrtc-codereview.appspot.com/735011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 08:53:55 +00:00
vspasova@webrtc.org
f61dc9be41 Moving frame_analyzer and rgba_to_i420_converter to src/tools. 
It might be useful to have these under src/tools as this way they will automatically sync in Chrome.

BUG=

Review URL: https://webrtc-codereview.appspot.com/740004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 08:12:00 +00:00
stefan@webrtc.org
2143c60315 Roll libvpx to 8bec177d. 
Also removing update_libvpx.sh as it is not needed when rolling libvpx
in WebRTC.

Verified on try bots.

BUG=

Review URL: https://webrtc-codereview.appspot.com/749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 07:53:21 +00:00
kma@webrtc.org
d81d906adf Fixed a bug in the C code in a filterbank function in iSAC-fix, and further optimized ARM code. 
The bug was introduced in the last CL during optimization.

With the new inine ARM assembly, the resulted assembly code of the loop is 28 instructions, compared to 47 lines by the original.

The optimized (with WEBRTC_ARCH_ARM_V7A defined in This file) and un-optimized code (with WEBRTC_ARCH_ARM_V7A not defined in This file) are bit-exact.
Review URL: https://webrtc-codereview.appspot.com/748004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-22 05:13:09 +00:00
leozwang@webrtc.org
d6fcf7f0da Add debug options to test app 
Description:
1. add apm debug option
2. add voice/video rtp dump option
3. front facing camera as the default, minor change

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/728012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 18:07:14 +00:00
vikasmarwaha@webrtc.org
bdb03d48ae Fix for issue 420 in TransmitMixer::SetTypingDetectionParameters. 
Review URL: https://webrtc-codereview.appspot.com/747004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 17:46:09 +00:00
leozwang@webrtc.org
510c1e36fe Read ptypes.txt from /sdcard on android 
BUG=
TEST=local
Review URL: https://webrtc-codereview.appspot.com/733011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 15:17:12 +00:00
stefan@webrtc.org
969d5f387b Reduce the PSNR threshold so that always having MFQE enabled doesn't trigger. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/743005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 11:33:29 +00:00
leozwang@webrtc.org
4ff8a9ad2f Print out more audio codec information in vie_auto_test 
It would be good to print more audio codec information to avoid
confusion

BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/744004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-21 03:50:24 +00:00
stefan@webrtc.org
4e8eabaab1 Properly handle switching between simulcast and unicast streams. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/733010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-20 14:29:52 +00:00
henrika@webrtc.org
3994e0324d ADM no longer reads PCM files from data/audio_device. Now uses the resource folder instead. 
BUG=737

Review URL: https://webrtc-codereview.appspot.com/714012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-20 09:44:59 +00:00
tina.legrand@webrtc.org
6f909835f1 Remove data files from data/audio_coding 
The files are already part of the resource package.

TBR=andrew@webrtc.org
BUG=issue737

Review URL: https://webrtc-codereview.appspot.com/724011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-20 07:39:52 +00:00
braveyao@webrtc.org
617e23cee6 In AVI file header, 'cbSize' might be omitted for PCM audio types. 
BUG = issue 671
TEST = Manual test
Review URL: https://webrtc-codereview.appspot.com/705012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-20 03:58:41 +00:00
henrike@webrtc.org
f7a58f868e Fixes VP8 issue with sending simulcast->non simulcast->simulcast. 
Review URL: https://webrtc-codereview.appspot.com/722013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 19:10:27 +00:00
mflodman@webrtc.org
6620c68b1a Changed test case for r2629. 
TBR=elham@webrtc.org

BUG=756

Review URL: https://webrtc-codereview.appspot.com/714011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-08-17 17:03:28 +00:00