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Author SHA1 Message Date
andrew@webrtc.org
6241beefa2 Refer to Chrome's DEPS to make rolling easier. 
- Sync Chrome's DEPS to chromium_deps/, and use the From() keyword
  to pull the correct revisions from it, rather than having to manually
  enter them.
- This idea is borrowed from the WebKit DEPS:
http://trac.webkit.org/browser/trunk/Source/WebKit/chromium/DEPS
- Fix update.py's DEPS parsing to handle From().
- Roll Chrome 120526:122775.
- Organize the deps alphabetically.
- Sync the in-tree gold linker, which is now required due to a change
  in the linker flags.
- Add the new deps to .gitignore.

BUG=
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/401004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 21:32:37 +00:00
marpan@webrtc.org
946601e408 Change default packetization mode to an equal size mode. 
This will produce equal size packets for each frame, which should be somewhat more favorable (less overhead/padding data) for the FEC.
Review URL: https://webrtc-codereview.appspot.com/396013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 18:52:53 +00:00
henrike@webrtc.org
70efc3250d Factory method for the ADM in the interface file. 
BUG=N/A
TEST=no

Review URL: https://webrtc-codereview.appspot.com/396017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 17:45:33 +00:00
phoglund@webrtc.org
dc3179dfe7 Implemented quick builds (e.g. only do full clean if the previous build failed). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/400009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 12:34:01 +00:00
xians@webrtc.org
6eb0ca2e75 Two problems are fixed: 
#1, avoid leaving the lock without entering the lock.
#2, race problems in variables like _playError, _recError, _recWarning, _playWarning.
Review URL: https://webrtc-codereview.appspot.com/400006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 10:39:53 +00:00
mflodman@webrtc.org
a556b0d193 Reverting r1749. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 10:15:04 +00:00
mflodman@webrtc.org
cb57f9ba95 Updated libyuv revision to include mjpg and added mjpg to type conversion. 
BUG=306
TEST=libyuv_unittests

Review URL: https://webrtc-codereview.appspot.com/407001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 09:47:07 +00:00
mflodman@webrtc.org
4f9e44f5c5 Prepared for MJPG capturing on Linux. MJPG is conversion is not available in libyuv yet, so this CL is only made as preparation. 
When this is available in libyuv, I'll remove the ifdef.

BUG=306
TEST=Manual loopback test with a high resolution, verify high FR.

Review URL: https://webrtc-codereview.appspot.com/397008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-23 09:00:26 +00:00
leozwang@webrtc.org
682cd4e9d1 Add android target 
Review URL: https://webrtc-codereview.appspot.com/396016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 16:05:19 +00:00
leozwang@webrtc.org
4ad4c24092 Add android to audio device module 
Review URL: https://webrtc-codereview.appspot.com/402001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 16:04:59 +00:00
kjellander@webrtc.org
c2e9cd3d37 Renaming windows slaves to follow naming convention better. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/402002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 15:38:40 +00:00
kjellander@webrtc.org
f25ab5d318 Enabling metrics_unittests that was created in http://webrtc-codereview.appspot.com/333025/ 
BUG=
TEST=Tested successfully using metrics_unittests in Debug+Release on Linux, Mac and Windows. No Valgrind warnings on Linux.

Review URL: https://webrtc-codereview.appspot.com/403001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 15:37:39 +00:00
stefan@webrtc.org
0fe2171b59 Relax libyuv test threshold and upgrade to libyuv r182. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=267
TEST=

Review URL: https://webrtc-codereview.appspot.com/391018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 11:21:18 +00:00
xians@webrtc.org
539ef94f20 Remove the deprecated kTraceModuleCall trace from audio coding module. 
Review URL: https://webrtc-codereview.appspot.com/399002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 08:35:03 +00:00
leozwang@webrtc.org
20e9cf274d Add android to video capture module 
Review URL: https://webrtc-codereview.appspot.com/399010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-22 00:40:45 +00:00
wu@webrtc.org
1181b31e47 Pull chromium version of libjingle and webrtc and build peerconnection sample server and client. 
Review URL: https://webrtc-codereview.appspot.com/399001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 23:44:42 +00:00
leozwang@webrtc.org
29fafefa0e Fix building errors 
Review URL: https://webrtc-codereview.appspot.com/399012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 19:46:33 +00:00
kjellander@webrtc.org
51198f1c68 More PRESUBMIT checks. 
Checks for:
- No iostream includes in headers
- No use of FRIEND_TEST for gtest
- Verifies that all C/C++ code passes cpplint.py check.
- Verifies that BUG= is present in commit message
- Verifies that TEST= is present in commit message

For more details, see Chrome's PRESUBMIT.py at
http://src.chromium.org/viewvc/chrome/trunk/src/PRESUBMIT.py?revision=113979&view=markup
and the canned checks at
http://src.chromium.org/viewvc/chrome/trunk/tools/depot_tools/presubmit_canned_checks.py?view=markup

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/317011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 17:53:46 +00:00
mallinath@webrtc.org
0d757b8610 Fixing coverity issues in capture module. 
Review URL: https://webrtc-codereview.appspot.com/399008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 16:47:55 +00:00
kjellander@webrtc.org
b9432cef96 Added simple wrapping for waterfall display. 
This is done by adding a whitespace before the last underscore on
strings above length 25, which should only affect the _unittest and
_integrationtest suffixes for our tests, that are the widest unwrappable
strings in our waterfall display.

I also removed all reduntant use of ShellCommand in favour for the
AddCommonStep function (that wraps the description text).

Also fixed so Mac tests are not halting the build.

Cleaned up some code style formatting.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/400005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 14:32:49 +00:00
niklas.enbom@webrtc.org
7cb0c240cb Trying to free up hellner from review work, since he mainly works in libJingle. 
Review URL: https://webrtc-codereview.appspot.com/392020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 13:58:58 +00:00
xians@webrtc.org
8435e8e3d8 Remove the deprecated kTraceModuleCall trace from audio processing module. 
Review URL: https://webrtc-codereview.appspot.com/399003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-21 10:37:26 +00:00
kjellander@webrtc.org
4e20a09686 Making buildbot output more compact. 
BUG=
TEST=Local master.

Review URL: https://webrtc-codereview.appspot.com/391002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 16:49:55 +00:00
phoglund@webrtc.org
3f6bf495d3 Fixed flunk settings: the builds show now halt only when compile and sync-kind operations fail. 
Fixed most flunk settings.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/401003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 15:21:44 +00:00
stefan@webrtc.org
a475556f5a Assume 200 ms RTT if we're only receiving. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/396012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 09:53:55 +00:00
xians@webrtc.org
20aabbb0be Remove the deprecated kTraceModuleCall trace from audio device module. 
Review URL: https://webrtc-codereview.appspot.com/396011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 09:17:41 +00:00
xians@webrtc.org
9a798d3fca Remove the deprecated kTraceModuleCall trace from video processing module. 
Review URL: https://webrtc-codereview.appspot.com/395012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 09:00:35 +00:00
stefan@webrtc.org
ca981180ee Upgrade libvpx to Duclair. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/400003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:55:19 +00:00
phoglund@webrtc.org
b45ceed9ef Rewrote the call report test. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/399006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:55:04 +00:00
xians@webrtc.org
843c8c78ff Remove the deprecated kTraceModuleCall trace from video modules. 
Review URL: https://webrtc-codereview.appspot.com/391015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:45:02 +00:00
xians@webrtc.org
6bde7a88f1 Remove the deprecated kTraceModuleCall trace from utility module. 
Review URL: https://webrtc-codereview.appspot.com/401002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:39:25 +00:00
xians@webrtc.org
57fb09ac18 Remove the deprecated kTraceModuleCall trace from udp transport module. 
Review URL: https://webrtc-codereview.appspot.com/395011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:38:21 +00:00
xians@webrtc.org
03039d56e6 Remove the deprecated kTraceModuleCall trace from media file module. 
Review URL: https://webrtc-codereview.appspot.com/392016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:37:49 +00:00
xians@webrtc.org
56cfe80c74 Remove the deprecated kTraceModuleCall trace from conference mixer. 
Review URL: https://webrtc-codereview.appspot.com/396010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-20 08:35:37 +00:00
tina.legrand@webrtc.org
145f04f0c4 Changing Celt to use stereo as default. 
Review URL: https://webrtc-codereview.appspot.com/397009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-18 00:32:16 +00:00
marpan@webrtc.org
bd5648f2db Reverting 1718: failed linux video test. 
TBR=stefan, andrew, marpan.
Review URL: https://webrtc-codereview.appspot.com/392018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 23:16:58 +00:00
marpan@webrtc.org
883e716304 Removed unused motion vector metrics from VideoContentMetrics; 
also removed other related unused variables and code. 

Reset frame rate estimate in mediaOpt when frame rate reduction is decided.

Update content_metrics with frame rate and qm_resolution with frame size.
Review URL: https://webrtc-codereview.appspot.com/395007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 18:35:23 +00:00
henrike@webrtc.org
f3760dc8e9 Fixes coverity warning that I missed in system wrappers. 
BUG=Coverity
TEST=N/A

Review URL: https://webrtc-codereview.appspot.com/395005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 16:27:25 +00:00
phoglund@webrtc.org
b3172860d7 Added a retry mechanism to vie_auto_test's verifying tests to make them less flaky. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/392015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 13:08:57 +00:00
mflodman@webrtc.org
4cb060127c Disabled RTPModule VP8 packetizer assert. 
BUG=293

Review URL: https://webrtc-codereview.appspot.com/399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 13:07:01 +00:00
phoglund@webrtc.org
8bfee84144 Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does. 
There are some small refactorings in here, but the real focus of this CL is in vie_autotest_rtp_fuzz.cc. This patch is mostly here to get a discussion going.

On my initial test the video engine doesn't recover, at least within 10 seconds of running with untampered packets. Not sure if this is according to specification though.

Ideas:
  - Generate random packets with correct RTP headers to get further into the code.
  - Don't generate fresh random data, but rather corrupt bits here and there in small amounts.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/383001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 09:32:48 +00:00
leozwang@webrtc.org
a52838b684 Update Android.mk and add test app 
Review URL: https://webrtc-codereview.appspot.com/388010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 01:16:43 +00:00
tina.legrand@webrtc.org
79e29e510f Adding option to change bitrate for Celt. 
I have updated the code so that Celt rate can be changed to any value between 48 and 128 kbps.
Tests for both mono and stereo are updated.Updated tests for Celt mono.

Review URL: https://webrtc-codereview.appspot.com/391014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-17 00:38:33 +00:00
wu@webrtc.org
133d1a18b7 Add a new folder so that we can pull webrtc and libjingle together and build peerconnection sample client and server. 
The DEPS file is mostly a placeholder right now.
Review URL: https://webrtc-codereview.appspot.com/390012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 21:51:35 +00:00
mallinath@webrtc.org
ee628358f4 Updating the object-c++ file after change in the API 
GetBestMatchedCapability
Review URL: https://webrtc-codereview.appspot.com/396009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 19:30:37 +00:00
mallinath@webrtc.org
8b4a98d0f4 Change in the interface file for GetBestMatchedCapability method. Updating mac files. 
Review URL: https://webrtc-codereview.appspot.com/389013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 19:00:28 +00:00
wu@webrtc.org
69f8be3875 Change the ExternalRenderer to provide both rtp timestamp and the render time. 
Review URL: https://webrtc-codereview.appspot.com/394006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:32:02 +00:00
mallinath@webrtc.org
12984f0d02 Fixing Coverity issues 
Note: This doesn't address Google Code style guidelines issues.
Review URL: https://webrtc-codereview.appspot.com/391011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:18:21 +00:00
xians@webrtc.org
3ab6dda5cb Truncated the volume to 255 when the users set the volume above 100%. 
Allowed the users to set the volume above 100% when AGC is enabled, in this case AGC can gradually scale down the volume instead of jumping to 100% immediately.
Reduced the flakiness of the volume tests in linux.
Review URL: https://webrtc-codereview.appspot.com/387011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 18:15:54 +00:00
mflodman@webrtc.org
f7b6078f6f Allow multiple send channels for REMB. Current implementation splits the remote estimate evenly between all senders. 
This CL will be followed by a CL adding support for several REMB groups.

TEST=video_engine_core_unittests

Review URL: https://webrtc-codereview.appspot.com/394002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:50:24 +00:00