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Author SHA1 Message Date
xians@webrtc.org
cf1b6aec30 iReduced the flakiness of the volume tests in linux pulseaudio 
Review URL: https://webrtc-codereview.appspot.com/390013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 17:22:49 +00:00
mflodman@webrtc.org
b4556cd29a Enabling mjpg for Windows. 
BUG=306
TEST=ViE loopback call on windows with resolution 960x720
Review URL: https://webrtc-codereview.appspot.com/411003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 14:02:12 +00:00
stefan@webrtc.org
1bb1da4c30 Enable MFQE if we are recieving temporal layers. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/411002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 13:52:34 +00:00
mflodman@webrtc.org
f3811194a5 Enable mjpg capture for Linux. 
BUG=306
TEST=ViE Loopback test using resolution larger than 640x480.

Review URL: https://webrtc-codereview.appspot.com/411001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 08:10:17 +00:00
mflodman@webrtc.org
8df260023b Prepared for MJPG capture without using MJPG DirectShow filter. MJPG is temporarily disabled and will enabled as soon as MJPG->I420 conversion is available. 
Review URL: https://webrtc-codereview.appspot.com/397011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-24 10:06:30 +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
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
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
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
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
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
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
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
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
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
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
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
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
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
stefan@webrtc.org
439be29445 Add APIs for getting receive-side estimated bandwidth and codec target rate. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 14:45:37 +00:00
braveyao@webrtc.org
590e5eb283 Convert audio layer to WAV on Vista RTM(without any Service Pack) 
Review URL: https://webrtc-codereview.appspot.com/397001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 03:21:05 +00:00
henrike@webrtc.org
d6d014ff12 Fixes memory leaks introduced in 1698. 
Review URL: https://webrtc-codereview.appspot.com/387014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-16 02:18:09 +00:00
henrike@webrtc.org
f5da4da409 Removes a global non POD instance from the RTP_RTCP module that was introduced in https://code.google.com/p/webrtc/source/detail?r=1076. 
Review URL: https://webrtc-codereview.appspot.com/314001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 23:54:59 +00:00
henrike@webrtc.org
05e0601160 Fixes coverity warnings in the udp_transport module. 
BUG=Coverity warnings.
TEST=N/A.

Review URL: https://webrtc-codereview.appspot.com/392012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 19:43:51 +00:00
henrike@webrtc.org
6b9253eb4f Fixe issues reported by Coverity for modules/utility. 
BUG=From Coverity
TEST=N/A

Review URL: https://webrtc-codereview.appspot.com/389011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 18:48:16 +00:00
henrike@webrtc.org
b38a66aaac Fixes a coverity warning in the mixer module. 
Review URL: https://webrtc-codereview.appspot.com/388009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-15 00:04:27 +00:00
marpan@webrtc.org
79a99de8e4 Reverting 1680: valgrind memory leak reported. 
TBR=marpan
Review URL: https://webrtc-codereview.appspot.com/392011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 22:37:10 +00:00
marpan@webrtc.org
738bcdc4ee Fix to coverity issue 10339. 
Review URL: https://webrtc-codereview.appspot.com/391010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 20:54:57 +00:00
andrew@webrtc.org
737c023e42 Properly disable sse2 source on non-x86. 
BUG=
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/387008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 19:57:50 +00:00
braveyao@webrtc.org
59d6cec291 Fix the crash at playing 48kHz stereo wav file. 
http://code.google.com/p/webrtc/issues/detail?id=208
Review URL: https://webrtc-codereview.appspot.com/396001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 18:17:16 +00:00
marpan@webrtc.org
4e34dcbd62 Allow for spatial-downsampling without reinitializaing encoder. Change of frame size will automatically trigger new key frame in codec. This feature is set off in vie_encoder until we upgrade to the new libvpx. 
Also reset frame rate estimate in mediaOpt when frame rate reduction is decided.
Review URL: https://webrtc-codereview.appspot.com/390006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 17:26:24 +00:00
mflodman@webrtc.org
d7d46887a6 Update receive only channels with RTT. 
vie_autotest_loopback.cc will not be part of the commit, it's only to show the test.

TEST=temporary with attached loopback test.

Review URL: https://webrtc-codereview.appspot.com/390007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-14 12:49:59 +00:00
pwestin@webrtc.org
c76c096c19 Bugfix issue 273, workaround for compiler issue. 
Review URL: https://webrtc-codereview.appspot.com/392005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-13 09:56:57 +00:00
pwestin@webrtc.org
52fd98d876 Removing encoder reset. Function did not make sence. 
Review URL: https://webrtc-codereview.appspot.com/391005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-13 09:03:53 +00:00
marpan@webrtc.org
567d507707 Fixes a bug when number of media packets in a frame is larger than maximum allowed for the generateFEC. 
Review URL: https://webrtc-codereview.appspot.com/391003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 18:56:14 +00:00
mflodman@webrtc.org
8224e19dd9 Fixed incorrect packet loss reported to encoder. 
BUG=275

Review URL: https://webrtc-codereview.appspot.com/394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:41:57 +00:00
pwestin@webrtc.org
5e954814a8 Clanup handling of key frame requests and FIR. 
Review URL: https://webrtc-codereview.appspot.com/387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-10 12:13:12 +00:00
andrew@webrtc.org
75f1948b0e Restore AECM Coverity fix. 
Add a test which would have caught the crash introduced by r1628.

BUG=274
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/388002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 17:16:18 +00:00