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816 Commits

Author SHA1 Message Date
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
stefan@webrtc.org
23307f7c4b Remove frame_list.cc from Andorid.mk. 
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 10:39:13 +00:00
tina.legrand@webrtc.org
df69775bfa Adding support for full-stereo codec. 
This is an experiment, letting Celt represent a full-stereo codec.

Review URL: https://webrtc-codereview.appspot.com/379013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 10:22:21 +00:00
stefan@webrtc.org
2979461595 Refactored the jitter buffer to use std::list. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/352016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 08:58:55 +00:00
stefan@webrtc.org
7dfa883954 Disable spatial subsampling for denoiser variance estimation. 
With subsampling there are sometimes quite visible trailing
artifacts.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/387002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 08:27:31 +00:00
pwestin@webrtc.org
95392e64ba Bugfix EnableIPV6 issue 255 
Review URL: https://webrtc-codereview.appspot.com/378005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 08:08:37 +00:00
kjellander@webrtc.org
1970b2fcb3 Fixing uninitialized codec settings struct in test. 
BUG=
TEST=video_codecs_test_framework_unittests passing in Debug+Release on Linux, Mac and Windows.

Review URL: https://webrtc-codereview.appspot.com/378004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 07:09:32 +00:00
andrew@webrtc.org
648af7423f Clean up MapSetting(). 
- Add assert(false) for "impossible" cases.
- Remove tests for invalid enum values.
- Modify MapError() to look the same way.

BUG=
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/386001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 01:57:29 +00:00
wu@webrtc.org
9143f774d1 Coverity fix for VideoRenderModule including issues 10084, 10226, 10267 and 10340. 
Review URL: https://webrtc-codereview.appspot.com/385001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 00:14:25 +00:00
bjornv@webrtc.org
236e842bca Removed memcpy of pointer to itself, triggering Valgrind warning. 
BUG=272
Review URL: https://webrtc-codereview.appspot.com/389002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 17:22:44 +00:00
phoglund@webrtc.org
78088c2f36 Removed warnings on Windows and enabled warnings-as-errors on Windows. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:56:45 +00:00
niklas.enbom@webrtc.org
87885e8409 This CL will look a bit strange. Essentially I've removed sanity checks for > 1 channel and then fixed the bugs that remained. Will add testing in a separate CL. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 14:48:59 +00:00
andrew@webrtc.org
daacee81b8 Use better reference files with audioproc_unittest. 
The files are shorter (7 s) with one set provided for each sample rate.

Will be accompanied by the following set of files in the resource bundle:
far8_stereo.pcm
far16_stereo.pcm
far32_stereo.pcm
near8_stereo.pcm
near16_stereo.pcm
near32_stereo.pcm

BUG=114
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/380003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 00:01:04 +00:00