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Author SHA1 Message Date
andrew@webrtc.org
56a0076d66 Add myself to the all_webrtc watchlist. 
Also fix documented_interface.

Review URL: https://webrtc-codereview.appspot.com/936022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 01:27:31 +00:00
fischman@webrtc.org
cb7561c945 Adding myself to webrtc watchlist. 
Review URL: https://webrtc-codereview.appspot.com/965023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 00:21:59 +00:00
leozwang@webrtc.org
6b9543b801 Add libpaced_sender to Android makefile 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/965022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 23:36:41 +00:00
leozwang@webrtc.org
d5fbdc8e52 Increase number of channels that can be supported on Android 
BUG=
TEST=local
Review URL: https://webrtc-codereview.appspot.com/967005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 21:30:34 +00:00
pwestin@webrtc.org
571a1c035b Enable paced sender. 
Review URL: https://webrtc-codereview.appspot.com/965016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 21:12:39 +00:00
stefan@webrtc.org
42aa10eba7 Clarifies the bandwidth estimation interfaces. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/965019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 15:02:13 +00:00
tina.legrand@webrtc.org
7577ddf27b Refactoring acm_generic_codec 
First patch: updating comments.

BUG=1024

Review URL: https://webrtc-codereview.appspot.com/936019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 14:09:35 +00:00
asapersson@webrtc.org
1726661ca2 Update parsed non ref frame info. 
Review URL: https://webrtc-codereview.appspot.com/932015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 13:39:51 +00:00
leozwang@webrtc.org
c734669dbf Re-create libvpx configuration files for Android 
Message:
I decided to separate libvpx configuraton files for linux/arm and
Android/arm. I will continue maintaining linux/arm configuration files.

The reasons to change:
It will be two sets of configuration files for android, one is arm, the
other one is arm-neon. For "arm" configuration, I will enable
"cpu-auto-detection" which means that binary can run on any armv7 devices.
For "arm-neon" configuration, it will be used for "neon" enabled devices
only. "arm" configuration will be the default configuration which is as
same as what we have for webrtc.

Description:
This CL separates libvpx configuration files for linux/arm and
Android/arm. Configuration files for Android arm are added, also
"cpu-auto-detection" is enabled.

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/934013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-13 01:19:07 +00:00
stefan@webrtc.org
8d18526834 Fixes an incorrect if statement in vie_sync_module.cc. 
BUG=1071

Review URL: https://webrtc-codereview.appspot.com/937018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 18:51:52 +00:00
thakis@chromium.org
6e470076ef mac: Fix a port leak in threading code. 
Chromium got the same fix in https://codereview.chromium.org/9169016/ 10 months ago.
Review URL: https://webrtc-codereview.appspot.com/929015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 18:30:59 +00:00
fischman@webrtc.org
b900721472 Fix OpenGL rendering of WebRTCDemo by accounting for stride != width. 
BUG=998

Review URL: https://webrtc-codereview.appspot.com/936021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 16:39:25 +00:00
henrik.lundin@webrtc.org
af49062a35 Revert 3071 - i420:verify image length 
This CL breaks vie_auto_test, test case
ViEVideoVerificationTest.RunsBaseStandardTestWithoutErrors.

BUG=

Review URL: https://webrtc-codereview.appspot.com/930016

TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/933014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-12 09:16:08 +00:00
fischman@webrtc.org
cb76c3c3f7 Unbreak ninja/android build of webrtc. 
Review URL: https://webrtc-codereview.appspot.com/932018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-10 22:39:37 +00:00
fischman@webrtc.org
6590ec3a62 Teach webrtc/codereview.settings how to point at svn rev's so rietveld issues get a useful URL. 
Review URL: https://webrtc-codereview.appspot.com/928021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-10 22:38:10 +00:00
fischman@webrtc.org
f4b26178df Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession. 
Review URL: https://webrtc-codereview.appspot.com/965015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 20:57:24 +00:00
pwestin@webrtc.org
b518017e71 Adding pacing module, will replace the transmission_bucket in the RTP module. 
TESTED=unittest
Review URL: https://webrtc-codereview.appspot.com/930015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 20:56:23 +00:00
mikhal@webrtc.org
f875fd22c0 i420:verify image length 
BUG=

Review URL: https://webrtc-codereview.appspot.com/930016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 18:48:30 +00:00
mikhal@webrtc.org
701567a864 Capture module: Fixing size computation for u and v planes 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/932017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 18:45:12 +00:00
leozwang@webrtc.org
06d72d881f Add Android OWNER files 
Message:
Add OWNER files so I can review and approve changes for Android.
I also should be owner for all .mk file, but it's OK for now,
please review.

BUG=None
TEST=None
Review URL: https://webrtc-codereview.appspot.com/932016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 17:51:55 +00:00
brykt@google.com
e8ef807a2d Added possibility to run quality modes test. Added possibility to input arguments to the test. The test will (for each frame) log the values in contentMetrics to a txt-file. The txt-file can optionally be saved in a specific place. Fixed an issue where video_coding_test crashed if there weren't any parameter submitted to an input argument. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/772005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 16:16:41 +00:00
phoglund@webrtc.org
9cb9fc17b1 Reformatted atomic32 files. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/937016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 08:57:25 +00:00
kma@webrtc.org
fa65c851fe Optimized function AllpassFilter2FixDec16() in isac fix for Android Neon platforms. 
With an offline test, codec cycles were reduced by 4%.
Review URL: https://webrtc-codereview.appspot.com/936007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-09 00:39:45 +00:00
tommi@webrtc.org
b952a90038 Remove an unused Shutdown method from the ThreadWrapper interface. 
The method was flagged by Chrome engineers as dubious since it uses
TerminateThread.  As it turns out, we don't use this method anywhere,
so we can simply remove it! :)

BUG=1066
Review URL: https://webrtc-codereview.appspot.com/938012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-08 13:40:59 +00:00
phoglund@webrtc.org
1401285fe7 Can now fully control custom calls from the command line. 
Example:
out/Debug/vie_auto_test --auto_custom_call --override "Enter destination IP.=127.0.10.55,Enter video send port.=12345"

The above will launch a call directly and choose all default values except for the overridden ones specified above.

BUG=

Review URL: https://webrtc-codereview.appspot.com/920008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-08 12:18:19 +00:00
wu@webrtc.org
2a749d3108 Verify output frame timestamp in VideoProcessingModuleTest.Resampler. 
TEST=unit tests
BUG=1069

Review URL: https://webrtc-codereview.appspot.com/964014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-08 01:07:21 +00:00
wu@webrtc.org
206532e6e2 Fix a bug in spatial_resampler where we should set the timestamp after Scale. 
TEST=try bots
BUG=1069

Review URL: https://webrtc-codereview.appspot.com/936018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 23:37:41 +00:00
kma@webrtc.org
12454028bc Fixed and enabled ARM assembly code in AECM and NS. 
Review URL: https://webrtc-codereview.appspot.com/860005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 22:34:31 +00:00
kma@webrtc.org
31eae47444 Implemented a build system that generates offset header files for ARM assembly files, in Android. 
The original CL was separated into two. Please refer to https://webrtc-codereview.appspot.com/860005 on how the build system and python script being used.
Review URL: https://webrtc-codereview.appspot.com/754005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 22:34:16 +00:00
mikhal@webrtc.org
055663be0a Updating vp8 tests 
Review URL: https://webrtc-codereview.appspot.com/936017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 21:36:15 +00:00
andrew@webrtc.org
c862f49b2d Move capture level computation after all processing. 
BUG=issue1065

Review URL: https://webrtc-codereview.appspot.com/930014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 19:08:03 +00:00
stefan@webrtc.org
7096fc0126 Break out unittest helpers for remote_bitrate_estimator. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/934012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 18:35:30 +00:00
mikhal@webrtc.org
ac993fef2c Adding codecType to OnIncomingCapturedEncodedFrame 
partially reverting r3013.

Review URL: https://webrtc-codereview.appspot.com/965010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 17:18:04 +00:00
pwestin@webrtc.org
c66e8b3f31 pre-factor cleanup pre-work. 
Review URL: https://webrtc-codereview.appspot.com/938010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 17:01:04 +00:00
phoglund@webrtc.org
4cebe6cded Made TickTime immutable, rewrote tick utils to be fakeable. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/798004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 13:37:19 +00:00
mflodman@webrtc.org
6e9890d1aa Removed ViEBaseObserver. 
BUG=1037
TEST=Still compiles and ViE autotest passes.

Review URL: https://webrtc-codereview.appspot.com/929012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 10:48:40 +00:00
kjellander@webrtc.org
8d0cef3bd2 Updating opus in .gitignore 
BUG=none
TEST=git status in a git-svn checkout.

Review URL: https://webrtc-codereview.appspot.com/965011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 08:22:10 +00:00
tina.legrand@webrtc.org
0ad3c1af0a Adding Opus stereo support to WebRTC 
This CL adds support for sending and receiving stereo using the Opus codec.

BUG=issue1013

Review URL: https://webrtc-codereview.appspot.com/930008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-07 08:07:29 +00:00
vikasmarwaha@webrtc.org
6dddfe9c35 Fix for webrtc issue 1052 on windows with vie_auto_test. 
Review URL: https://webrtc-codereview.appspot.com/929014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 21:19:52 +00:00
andrew@webrtc.org
ddcc9429e7 Check the channels in receive-side processing frames. 
The number of channels must be set correctly before calling ProcessStream. This
was preventing stereo frames from being processed.

Also fix voe_cmd_test, which wasn't enabling rx NS properly.

BUG=issue713, 7375579

Review URL: https://webrtc-codereview.appspot.com/929013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 18:39:40 +00:00
asapersson@webrtc.org
e5b49a0472 Update timestamp offset for re-transmitted packets. 
BUG=1059
Review URL: https://webrtc-codereview.appspot.com/930011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-06 13:09:39 +00:00
kjellander@webrtc.org
f17d7d1dde Using proper GYP references for Strmiids.lib on Windows 
This makes it possible to link properly using Ninja on Windows.

Another issue exists due to the path containing spaces in webrtc/modules/video_capture/windows/direct_show_base_classes.gyp but that can be worked around by copying the baseclasses dir from the SDK into a path without spaces and then overriding the direct_show_dir variable with GYP_DEFINES.

Example:

set GYP_DEFINES=direct_show_dir=C:/WinSDKv7.1_directshow_baseclasses/
gclient runhooks
ninja -C out/Debug

Notice that the ending slash is needed for the direct_show_dir variable.

BUG=none
TEST=local compilation with the baseclasses copied from Windows SDK 7.1 into C:/WinSDKv7.1_directshow_baseclasses.

Review URL: https://webrtc-codereview.appspot.com/936014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 13:13:05 +00:00
tina.legrand@webrtc.org
f7fa6276e2 Reformating files in audio coding module. 
This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-05 09:35:51 +00:00
kjellander@webrtc.org
a56d759723 Removing use of raw buffers for I420PSNR and I420SSIM functions 
BUG=none
TEST=metrics_unittest, video_codecs_test_framework_integrationtests in normal and memcheck mode.

Review URL: https://webrtc-codereview.appspot.com/937013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-02 22:36:14 +00:00
leozwang@webrtc.org
5f9970fb0b Refactor OpenSL audio driver 
Message:
I want to start this review, the basica framework is almost done.

Description:
This implementation is similar to current one, but
1. followed the design doc at
https://docs.google.com/a/google.com/document/d/1g5q2SVtkFPl2OSjvSF3eeLb_S7sCnbINxiBUES6XLCM/edit
which uses two threads, playout thread and recording thread, uses large
audio buffer, etc.
2. google code style.

What are missing in this cl,
1. a better way to control schedule/thread priority.
2. java/jni interface to better support what cannot be done in OpenSL.

Please take a review, thanks.
Review URL: https://webrtc-codereview.appspot.com/902004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-02 16:48:03 +00:00
mikhal@webrtc.org
737ed3bfa5 libyuv wrapper: 1. Updating rotation settings - in case of 90 or 270 degree rotations, width and height should be updated accordingly. 2. Test clean-up. 
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/936008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-01 15:45:38 +00:00
mflodman@webrtc.org
1be46fc721 Change src/webrtc in WATCHLIST. 
TEST=ViE people were auto-cc on this dummy cl http://review.webrtc.org/938007/

Review URL: https://webrtc-codereview.appspot.com/939010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-01 08:09:09 +00:00
kjellander@webrtc.org
f3ffcceaf8 Adding default trybot names to PRESUBMIT.py. 
With this change, only the listed trybots will be the ones that a tryjob
is sent to. This works because there's a check in depot_tools/presubmit_support.py that looks for a function defined with the name 'GetPreferredTrySlaves'.

This makes it possible for us to add new trybots that will not receive jobs by default (only when the --bot flag is specified).

This CL is needed before we add the following try bots:
linux_memcheck
linux_tsan
linux_asan

BUG=none
TEST=submitting jobs to a local try server, with and without the --bot
flag.

Review URL: https://webrtc-codereview.appspot.com/939008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 14:52:21 +00:00
niklas.enbom@webrtc.org
ef629299e9 Landing http://review.webrtc.org/914006/ 
Review URL: https://webrtc-codereview.appspot.com/930007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 14:35:11 +00:00
stefan@webrtc.org
1a2a6dda26 Fixes a bitrate mismatch between sender and receiver. 
TEST=trybots

BUG=

Review URL: https://webrtc-codereview.appspot.com/928014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-31 12:21:13 +00:00