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928 Commits

Author SHA1 Message Date
kjellander@webrtc.org
2050f84b98 audio_device_test_api failing cleaner failure for Linux without audio devices. 
BUG=None
TEST=audio_device_test_api on Linux.

Review URL: https://webrtc-codereview.appspot.com/447002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-09 14:22:27 +00:00
tina.legrand@webrtc.org
0dab9e1523 Revert of r1859 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-09 10:03:09 +00:00
henrika@webrtc.org
907bc55c19 Removes WebRtc_Word8 dependecy in the AudioDeviceModule. 
This CL also modifies the ADM callback interface and introduces void* instead of WebRtc_Word8*
as pointer types for data buffers. This change also affects the VoiceEngine.
Review URL: https://webrtc-codereview.appspot.com/443001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-09 08:59:19 +00:00
kjellander@webrtc.org
67fdd70e1b Refactoring audio_device_test_api for gtest and execution on all platforms. 
All the code that was previously in one single function is now broken up into 44 gtest tests. The creation of the Audio Device is done once (SetUpTestCase) and the audio device is initialized before each test (SetUp) and terminated after each test (TearDown). Doing this, the things that execute are basically the same since the test was structured as different sections separated by these calls:
TEST(audioDevice->Terminate() == 0);
TEST(audioDevice->Init() == 0);

I also cleaned up some unused helper functions and removed old test macros when replacing them by gtest macros.

The parts that are failing for Mac and Windows are excluded using #ifdef. Separate issues are filed for
this code to be fixed.

Formatting is updated to follow the WebRTC style guide.

BUG=None.
TEST=audio_device_test_api in Debug+Release on Linux, Mac and Windows. Test run audio_device_test_func on Linux.

Review URL: https://webrtc-codereview.appspot.com/437002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-09 08:11:04 +00:00
tina.legrand@webrtc.org
f1befad273 Reregister of stereo receiver didn't work. 
This CL takes care of the re-registration of codecs, and tests unregistering stereo codecs.

One bug fixed in Celt too.

TEST=audio_coding_module_test: TestStereo.

Review URL: https://webrtc-codereview.appspot.com/436002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-08 13:23:18 +00:00
mflodman@webrtc.org
7845d07bf8 VideoCapture now uses pointer constructor of CriticalSectionScoped. 
BUG=184
TEST=video_capture_module compiles on all platforms when removing ref ctor of CriticalSectionScoped.

Review URL: https://webrtc-codereview.appspot.com/434001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-08 08:09:17 +00:00
marpan@webrtc.org
accf607b3e Updates for resolution adaptation. 
1) added support for two additional modes: 
    -3/4 spatial down-sampling
    -2/3 frame rate reduction
2) updated unittest and added a few more tests
3) some code refactoring
Review URL: https://webrtc-codereview.appspot.com/429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-07 17:16:10 +00:00
leozwang@webrtc.org
57da718734 Fix building errors on android 
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/441001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 21:28:48 +00:00
leozwang@webrtc.org
77fe431f57 Enable video render test on android 
Review URL: https://webrtc-codereview.appspot.com/428006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:55 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type 
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 20:59:13 +00:00
andrew@webrtc.org
6f9f817e06 Add an API to offset system delay. 
Plumb it through VoiceEngine.

BUG=
TEST=voe_auto_test,audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/428010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 19:03:39 +00:00
mflodman@webrtc.org
0e703f4d0d VideoRender now uses pointer constructor of CriticalSectionScoped. 
BUG=184
TEST=video_render_module compiles on all platforms when removing ref ctor of
CriticalSectionScoped.

Review URL: https://webrtc-codereview.appspot.com/427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-06 12:02:20 +00:00
leozwang@webrtc.org
db2de5b49f Fix building errors on android 
TBR=Tina

BUG=
TEST=build on android
Review URL: https://webrtc-codereview.appspot.com/430001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 19:53:24 +00:00
leozwang@webrtc.org
66487e1629 Enable video test on android 
Review URL: https://webrtc-codereview.appspot.com/429006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 19:34:06 +00:00
mflodman@webrtc.org
9ec883e8bd Allow multiple REMB groups and introduce receive channels. 
BUG=312
TEST=ViE standard autotest and API test.

Review URL: https://webrtc-codereview.appspot.com/432005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 17:12:41 +00:00
leozwang@webrtc.org
855ced7336 Further cleanup WebRtc_Word8 
Review URL: https://webrtc-codereview.appspot.com/426008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 16:07:51 +00:00
mflodman@webrtc.org
fa6bc673b0 Changed default module condition for BW estimate. 
Review URL: https://webrtc-codereview.appspot.com/433001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-05 11:59:55 +00:00
leozwang@webrtc.org
42e362eee5 Fix compilation error on android 
Review URL: https://webrtc-codereview.appspot.com/426006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-02 17:14:09 +00:00
leozwang@webrtc.org
3197d48407 Enable audio device test on android 
Review URL: https://webrtc-codereview.appspot.com/428005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-02 17:12:14 +00:00
marpan@webrtc.org
26762e3e40 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 video engine until we upgrade to a newer version of libvpx.
Review URL: https://webrtc-codereview.appspot.com/427003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-02 16:48:36 +00:00
leozwang@webrtc.org
fa8c9f7a4f Remove unused variable 
Review URL: https://webrtc-codereview.appspot.com/432003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-02 07:15:03 +00:00
leozwang@webrtc.org
c9a3b81fd2 Further cleanup WebRtc_Word8 in video_capture on mac 
BUG=311
TBR=Wu, Mallinath
Review URL: https://webrtc-codereview.appspot.com/431002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 19:59:52 +00:00
leozwang@webrtc.org
4add6bc603 Fix building errors on window which caused by previous cl 
BUG=311
TBR=Wu, Mallinath
Review URL: https://webrtc-codereview.appspot.com/432002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 19:57:13 +00:00
leozwang@webrtc.org
09e771998c Correct WebRtc_word8 usage in media file module 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/427002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 18:35:54 +00:00
leozwang@webrtc.org
28f3913ca9 Correct WebRtc_Word8 in adm 
Correct WebRtc_Word8 usage in adm

BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=buidl on all platforms

Review URL: https://webrtc-codereview.appspot.com/428001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 18:01:48 +00:00
leozwang@webrtc.org
0689271d64 nits 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 16:31:00 +00:00
leozwang@webrtc.org
1745e932cc Correct wrong usage of WebRtc_Word8 in video capture 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 16:30:40 +00:00
tina.legrand@webrtc.org
1f2cabaecd Crash when deleting Celt. 
BUG=issue 6087770
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/420001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-03-01 08:46:09 +00:00
kjellander@webrtc.org
132eccbb69 Renamed platform specific code to use GYP conventions. 
Restructured GYP files a bit to clean up things.
Removed copying of images to /tmp
Fixed output location of DumpFileName.rtp.

BUG=None
TEST=Tested compiling and running on Mac, Win, Linux.

Review URL: https://webrtc-codereview.appspot.com/406002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 20:55:25 +00:00
leozwang@webrtc.org
07c68b9c9d Correct wrong usage of WebRtc_Word8 in rtp and udp module 
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/418001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 16:09:51 +00:00
marpan@webrtc.org
4788bf4256 Fix to warnings on windows. 
Review URL: https://webrtc-codereview.appspot.com/415004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-29 01:11:39 +00:00
marpan@webrtc.org
9d76b4ea54 Updates for resolution adaptation: 
1) code cleanup and some updates to selection logic for qm_select.
2) added unit test for the QmResolution class.
3) update codec frame size and reset/update frame rate in media-opt:
4) removed unused motion vector metrics and some related code of content metrics processing.
Review URL: https://webrtc-codereview.appspot.com/405008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-28 23:39:31 +00:00
andrew@webrtc.org
547c157a49 Temporarily use _Word8 to avoid clang error. 
BUG=issue311
TEST=build on clang

Review URL: https://webrtc-codereview.appspot.com/415003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-28 22:30:30 +00:00
leozwang@webrtc.org
91b359ea9b Change WebRtc_Word8 to char 
Review URL: https://webrtc-codereview.appspot.com/407003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-28 17:26:14 +00:00
stefan@webrtc.org
4ce0ba00de Fix issue 310. 
BUG=310
TEST=session_info_unittest.cc

Review URL: https://webrtc-codereview.appspot.com/404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-28 12:09:09 +00:00
henrike@webrtc.org
26085e18e0 Coverity fixes for module/media_file. 
BUG=Coverity report.
TEST=N/A.

Review URL: https://webrtc-codereview.appspot.com/397003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 21:50:40 +00:00
leozwang@webrtc.org
ead7d25c1a Revert r1775 which caused building errors. 
TBR=pwestin@webrtc.org, mflodman@webrtc.org

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 19:45:09 +00:00
leozwang@webrtc.org
2559cbf7b7 Change WebRtc_Word8 to char 
Review URL: https://webrtc-codereview.appspot.com/405003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 19:18:25 +00:00
leozwang@webrtc.org
3e9e0f0497 Change WebRtc_Word8 to char 
Review URL: https://webrtc-codereview.appspot.com/405004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 19:17:38 +00:00
leozwang@webrtc.org
adb89f56e0 Change WebRtc_Word8 to char 
Review URL: https://webrtc-codereview.appspot.com/405005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-27 19:12:19 +00:00
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
stefan@webrtc.org
4b377414f1 Fix release build errors. 
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/394005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 13:50:57 +00:00
xians@webrtc.org
3dbed8597e This CL makes the playout delay value thread safe. 
With the patch, _sndCardPlayDelay is calculated in the DoRenderThread instead of capture thread, an capture thread only gets the _sndCardPlayDelay value.
And _sndCardPlayDelay and _sndCardRecDelay are only changed to be Atomic32 to make them to be accessed by multiple threads.


Test=None
Bug=256
Review URL: https://webrtc-codereview.appspot.com/394001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 13:50:48 +00:00
stefan@webrtc.org
9c84b0dc9f Fix build errors with GCC. 
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 13:14:04 +00:00
stefan@webrtc.org
7adab0922d This removes the knowledge of frame completeness from the FEC decoder. 
Therefore, with this change a recovered packet is only considered old,
and will be removed, if more than 48 recovered packets are stored.

Packets are immediately reconstructed and sent to the jitter
buffer. Before this CL packets were processed on a frame-by-frame
basis, and all packets belonging to a frame was sent to the
jitter buffer at the same time.

The number of FEC packets is also limited to 48. An FEC packet is
removed if all protected packets have been recovered or if the
FEC packet is considered old.

Lot's of tests added.

Patch-set 2:
- Fixed rtp_fec_unittest.cc to work with the new reconstruction.
- Added reference counting of Packet to be able to keep references to them from FecPacket between different reconstruction runs.
- Rewrote the packet search to use std::set_intersection.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 12:34:52 +00:00
kjellander@webrtc.org
cf6a295b13 Making video codecs test framework integration test execute in a reproducable fashion. 
Fixed reproducable random behavior in packet_manipulator.h.
Test is now fully reproducable (runs on only one core) so much tighter limits are now set for the SSIM/PSNR values for the encoding/decoding (verified on all platforms)

BUG=
TEST=out/Debug/video_codecs_test_framework_integrationtests in Debug+Release on Linux, Mac, Windows and in Linux Valgrind.

Review URL: https://webrtc-codereview.appspot.com/381005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-09 09:01:51 +00:00
henrike@webrtc.org
d5657c2f69 Refactored files according to google style since http://review.webrtc.org/314001/ is blocked on this and formatting changes should not be done with code changes. 
Review URL: https://webrtc-codereview.appspot.com/387005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 23:41:49 +00:00
andrew@webrtc.org
68da6adafe Remove WebRtc_ types. 
Allows us to avoid the "cast to UWord32" Coverity coverup.

BUG=
TEST=test_fec

Review URL: https://webrtc-codereview.appspot.com/379002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 22:24:14 +00:00
wu@webrtc.org
a8084b07e3 Revert r1628 which causes the crash of voe_auto_test. 
With r1628, it's possible the second memcpy got a NULL nearendClean.

TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/390005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 17:56:39 +00:00
tina.legrand@webrtc.org
13ac430bef Adding missing timestamp calculation to RTPencode. 
Review URL: https://webrtc-codereview.appspot.com/392002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 13:20:36 +00:00
mflodman@webrtc.org
d2940f71e4 VCM::JB critsect fix. 
Review URL: https://webrtc-codereview.appspot.com/390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 12:42:56 +00:00