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977 Commits

Author SHA1 Message Date
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
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
wu@webrtc.org
50099af75f Disable flaky test VideoProcessorIntegrationTest.Process5PercentPacketLoss. 
BUG=262
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/379014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 22:50:48 +00:00
marpan@webrtc.org
6584e58001 Coverity fix for issues 10325,10326. 
Review URL: https://webrtc-codereview.appspot.com/377001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 19:02:54 +00:00
wu@webrtc.org
13e0345b35 Fix uninitialized variable error in Relase mode. 
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/377007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 16:19:15 +00:00
mflodman@webrtc.org
517e5e3846 NetEQ switch fix. 
Review URL: https://webrtc-codereview.appspot.com/381006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 15:04:00 +00:00
stefan@webrtc.org
94355e0a59 Fix crash in SessionInfo::BuildSoftNackList. 
BUG=259
TEST=

Review URL: https://webrtc-codereview.appspot.com/377006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 14:06:39 +00:00
mflodman@webrtc.org
a39621ee1b Disabling APM test for invalid enum values. 
Review URL: https://webrtc-codereview.appspot.com/378006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 14:00:12 +00:00
mflodman@webrtc.org
ec31bc1321 Fixed APM tests. 
TEST=ApmTest.*

Review URL: https://webrtc-codereview.appspot.com/380008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 12:42:45 +00:00
mflodman@webrtc.org
657b2a4965 Added return due to gcc complaints in r1604. 
TBR=andrew

TEST=Bulid with clang version 3.1 (trunk 148911) and gcc.

Review URL: https://webrtc-codereview.appspot.com/384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 11:06:01 +00:00
mflodman@webrtc.org
c80d9d9361 Removed default cases causing clang errors, -Wcovered-switch-default. 
BUG=
TEST=Bulid with clang version 3.1 (trunk 148911)

Review URL: https://webrtc-codereview.appspot.com/379008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-06 10:11:25 +00:00
andrew@webrtc.org
4942832928 Fix "may be used uninitialized" warning. 
TBR=marpan@webrtc.org
BUG=
TEST=build on Linux/Release and rtp_rtcp_unittests

Review URL: https://webrtc-codereview.appspot.com/381004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 05:23:51 +00:00
marpan@webrtc.org
b783a55df3 Unit test for forward_error_correction. 
Review URL: https://webrtc-codereview.appspot.com/358006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 02:46:35 +00:00
marpan@webrtc.org
307c1ff20c Fix for issue #254: windows crash of test_fec. 
Review URL: https://webrtc-codereview.appspot.com/379010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-04 02:45:22 +00:00
andrew@webrtc.org
dde977ec83 AudioFrame payload shouldn't be mutable. 
This requires making Mute() non-const, which is correct anyway.

BUG=
TEST=voe_auto_test on Linux

Review URL: https://webrtc-codereview.appspot.com/376001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-03 17:47:32 +00:00
henrik.lundin@webrtc.org
683833442a Fix for warning in GCC 4.6 
Upstream copy of a fix provided in http://codereview.chromium.org/9309007/.

BUG=none
TEST=neteq_unittests

Review URL: https://webrtc-codereview.appspot.com/383002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-03 12:33:50 +00:00
henrik.lundin@webrtc.org
82e1c8d0e7 Fix for issue 253 
Initializing a few arrays to avoid compiler warnings under
the O3 flag.

BUG=http://code.google.com/p/webrtc/issues/detail?id=253

Review URL: https://webrtc-codereview.appspot.com/380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-03 09:46:01 +00:00
pwestin@webrtc.org
fdf21c8c55 Removed dead version code. 
Review URL: https://webrtc-codereview.appspot.com/377003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-02 12:46:58 +00:00
pwestin@webrtc.org
4ea57e5e26 Changed VP8 to follow the style guide a little bit more. 
Review URL: https://webrtc-codereview.appspot.com/379003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-02 12:21:47 +00:00
stefan@webrtc.org
07b45a5dc4 Added API for getting the send-side estimated bandwidth. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/372006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-02 08:37:48 +00:00
kma@webrtc.org
de66b91274 In voice engine, added member audioFrame to classes AudioCodingModuleImpl and VoEBaseImpl, 
and touched VoEBaseImpl::NeedMorePlayData and AudioCodingModuleImpl::PlayoutData10Ms(), for
performance reasons in Android platforms.
The two functions used about 6% of VoE originally. After the change, the percentage reduced
to about 0.2%.
Review URL: https://webrtc-codereview.appspot.com/379001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 18:39:44 +00:00
andrew@webrtc.org
7fe219f681 Add some additional checks for corrupt payload. 
Investigation with corrupt payloads revealed a few places we could
be using stronger checks. These are not foolproof by any means, but
I figure the earlier we catch this the better.

BUG=242
TEST=loopback call with a hacked ViE to insert corrupt payloads, and vie_auto_test without the hacks.

Review URL: https://webrtc-codereview.appspot.com/369015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 02:40:37 +00:00
kma@webrtc.org
727a0a03a1 Fixed a bug in assembly code in aecm_core.c (hasn't caused a problem yet). 
Did apm unit test. Bit exact.
Review URL: https://webrtc-codereview.appspot.com/366010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-01 00:05:22 +00:00
frkoenig@google.com
d8f58a4ab0 Cross platform build fix for SSIM (part 2) 
Data alignment fix for SSIM.

WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128.  By declaring the 
variable as __m128i it will always be 128 bytes aligned.

Related to issue 239013.
http://webrtc-codereview.appspot.com/239013/
Review URL: https://webrtc-codereview.appspot.com/375004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 17:49:38 +00:00
henrik.lundin@webrtc.org
dd478e2081 Fix for warning in GCC 4.6 
Upstream copy of a fix provided in http://codereview.chromium.org/9159058/.

Review URL: https://webrtc-codereview.appspot.com/369024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 13:12:41 +00:00
stefan@webrtc.org
91c630851a Fix potential VCMReceiver crash. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/368012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 10:49:08 +00:00
marpan@webrtc.org
cdba1a836b test_fec: Reduce execution time of test, and use testsupport/fileutils.h for path of randomSeedLog file. 
Review URL: https://webrtc-codereview.appspot.com/373016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-31 00:36:14 +00:00
andrew@webrtc.org
293d22b39b Add a new macro for bit-exact audioproc tests. 
Enable bit-exact test for all fixed-point configs.

BUG=114
TEST=audioproc_unittest on all platforms.

Review URL: https://webrtc-codereview.appspot.com/369018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 22:04:26 +00:00
andrew@webrtc.org
40654039cd Use pointer-based CriticalSectionScoped(). 
The reference-based constructor is deprecated.

BUG=185
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/373015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 20:51:15 +00:00
kma@webrtc.org
89a100092a A minor change in function WebRtcNetEQ_PacketBufferFindLowestTimestamp for 
NetEq, for performance reasons.
In Android platform, with an offline testing file, the function cycles was reduced by 25%.
This function was also reformatted.
Review URL: https://webrtc-codereview.appspot.com/367010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 15:37:33 +00:00
pwestin@webrtc.org
5dad00be52 Coverty fix: FEC unintended signed extension and resource leaks. 
Review URL: https://webrtc-codereview.appspot.com/368010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1569 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 13:05:29 +00:00
mflodman@webrtc.org
d3b22c9356 Resolved X11 shared memiory leak. 
BUG=248
TEST=See bug

Review URL: https://webrtc-codereview.appspot.com/367016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 09:44:28 +00:00
bjornv@webrtc.org
0c6f931420 Removed versions in module/audio_processing and common_audio/vad. 
Affected vad_unittest only.
In addition changed to correct header guards.
Review URL: https://webrtc-codereview.appspot.com/367019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 09:39:08 +00:00
stefan@webrtc.org
2fd1e1e40d Add unittests for ReceiverFec. 
Also added mock for RtpReceiverVideo and did appropriate changes to
allow for mocking.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-30 09:03:37 +00:00
pwestin@webrtc.org
04cf69a714 Coverty: cleanup CheckCSRC. 
Review URL: https://webrtc-codereview.appspot.com/369014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 13:47:19 +00:00
phoglund@webrtc.org
2f7740973d Fixed C errors from GCC 4.6. 
Fixed errors in .c files.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/373014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 13:44:26 +00:00
mflodman@webrtc.org
1f992807eb Fixed frame scaler bugs. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 13:42:53 +00:00
tina.legrand@webrtc.org
cbe1de9aa6 This CL solves three remaining Coverity warnings. 
A few more members were left uninitialized, and one more size mismatch in a multiplication.

Review URL: https://webrtc-codereview.appspot.com/367001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-27 09:00:46 +00:00
mallinath@webrtc.org
a8c568f0a4 Fix external codec erase in destructor. 
Review URL: https://webrtc-codereview.appspot.com/368008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-26 16:19:03 +00:00
phoglund@webrtc.org
d1a860b415 Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans. 
Fixed remaining compilation errors in release, etc.

Fixed errors from GCC 4.6 compilation.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/366008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-26 14:49:28 +00:00
andrew@webrtc.org
42ae41e5a2 Fix enumeral comparison error. 
TBR=henrike
BUG=
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/372007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 19:38:16 +00:00
andrew@webrtc.org
b9d7d934de Rename interface/ to include/ in audio_processing. 
BUG=none
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/367007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 19:21:13 +00:00
andrew@webrtc.org
24bd58e689 Properly count anonymous mixing participants. 
When _amountOfMixableParticipants == 1, we skip mixing and saturation
protection. Without this fix, an anonymous participant would only be
properly counted if it was the last added.

For example, if an anonymous participant was added first, followed by
a regular participant, _amoutOfMixableParticipants would == 1 and the
regular participant would not be mixed.

BUG=issue209
TEST=New test added to voe_auto_test to verify, and used voe_cmd_test.

Review URL: https://webrtc-codereview.appspot.com/367006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 18:57:44 +00:00
henrik.lundin@webrtc.org
dcf006480c Fix typo in a comment 
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/369012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1548 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 16:48:00 +00:00
henrik.lundin@webrtc.org
4679652d57 Implemented a fix for Issue 88. 
NetEQ now checks for too early CNG packets, and modifies the CNG
sample counter to jump forward in time if needed to combat clock
drift.

Adding a new unittest to reproduce and solve the issue. The
unittest LongCngWithClockDrift verifies that the buffer delay
before and after a long CNG period is almost constant. The test
introduces a clock drift of 25 ms/s.

BUG=http://code.google.com/p/webrtc/issues/detail?id=88
TEST=neteq_unittests NetEqDecodingTest.LongCngWithClockDrift

Review URL: https://webrtc-codereview.appspot.com/372002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1547 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 16:37:41 +00:00
bjornv@webrtc.org
f4b77fd722 VAD refactor: Mode changed to "int". 
As part of style this CL includes changing the input aggressiveness mode from int16_t to int. No other style changes made.
Impact on:
- Audio Processing: Changed return value on MapSetting().
- Function test in audio_conference_mixer already uses int. No action.
- NetEq: Function pointer changes and input parameter changes in SetVADMode() and SetVADModeInternal().
- Audio Coding: Uses enum ACMVADMode which is type independent.
- VAD: Two unit tests.

TESTS=vad_unittests, neteq_unittests, audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/373001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-25 12:40:00 +00:00
henrike@webrtc.org
567b99be5f Coverity report: fixes an issue where the returnvalue of a function is not checked. 
Review URL: https://webrtc-codereview.appspot.com/347013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 23:43:54 +00:00
andrew@webrtc.org
f5d8c3bc3b Fix audioproc_unittest on Windows. 
On Windows, files have to be closed before they can be removed.

TBR=bjornv
BUG=none
TEST=audioproc_unittest on Windows.

Review URL: https://webrtc-codereview.appspot.com/369010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 21:35:39 +00:00
pwestin@webrtc.org
f6bb77a6f0 Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace. 
Review URL: https://webrtc-codereview.appspot.com/358001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 17:16:59 +00:00
mallinath@webrtc.org
218db3de20 Iterator was invalid while removing entries from codec db maps. 
Review URL: http://webrtc-codereview.appspot.com/373003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 17:11:44 +00:00
stefan@webrtc.org
9e332ab95b Make sure we check the return value from shmat(). 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/358007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 16:33:27 +00:00
pwestin@webrtc.org
b73c3d1f5d Bugfix android build. 
Review URL: https://webrtc-codereview.appspot.com/374003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 15:25:30 +00:00
pwestin@webrtc.org
28a5cb29ab Bugfix receive side only packet loss estimate with NACK. 
Review URL: https://webrtc-codereview.appspot.com/373006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 14:34:58 +00:00
punyabrata@webrtc.org
6da8eeb946 Removing an assert for a case that can occur 
when corrupt packets are injected into voice engine.
Review URL: https://webrtc-codereview.appspot.com/373004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-24 00:48:36 +00:00
leozwang@webrtc.org
f9cd693145 Enable vp8 and videoengine on android 
Review URL: https://webrtc-codereview.appspot.com/368003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:56:13 +00:00
leozwang@webrtc.org
a45d05a341 Add brighten.cc to makefile 
Review URL: https://webrtc-codereview.appspot.com/369003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:54:13 +00:00
leozwang@webrtc.org
376be6c904 Fix compilation error 
Review URL: https://webrtc-codereview.appspot.com/358005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:46:38 +00:00
pwestin@webrtc.org
b30f0edce6 Bugfix buffer usage out of scope. 
Review URL: https://webrtc-codereview.appspot.com/372001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 16:23:31 +00:00
stefan@webrtc.org
175fecde97 Fix clang build error. 
TBR=henrik.lundin@webrtc.org
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 15:23:31 +00:00
stefan@webrtc.org
8fe03af674 Refactor to use std::list in the video rtp play tools. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/349013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 14:56:14 +00:00
bjornv@webrtc.org
152c34cf11 VAD-refactor. Changed to int as return value for WebRtcVad_set_mode(). 
Impact on NetEq function pointers. Other components already treat the output as int. These are:
* audio_processing
* funtion test in audio_conference_mixer
* audio_coding

TEST=vad_unittests, neteq_unittests
Review URL: https://webrtc-codereview.appspot.com/367003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-23 12:36:46 +00:00
andrew@webrtc.org
e2ed5baf47 Enable audioproc_unittest on all platforms. 
But, for the time being, limit the bit-exact test to 64-bit Linux debug.

TEST=build and run all configs on Linux, and standard configs on Win and Mac.

Review URL: https://webrtc-codereview.appspot.com/343005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 19:06:38 +00:00
stefan@webrtc.org
f27916a76a Remove use of MapWrapper in video_coding. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/344018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 14:04:13 +00:00
henrik.lundin@webrtc.org
d798953846 NetEqRTPplay modification 
Make the program look for the ptypes.txt file in the default trunk
path, if the path to the executable indicates that it sits in the
trunk/out/Debug folder.

Changing PT for CNG-WB to 98

Remove warnings when building NetEQ with NETEQ_DELAY_LOGGING

Review URL: https://webrtc-codereview.appspot.com/339003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 13:42:16 +00:00
kjellander@webrtc.org
5e1625ed2d Fixing Valgrind problem detected by video_processing_unittests. 
Simple initialization of the allocated memory for the image buffer avoids reading uninitialized data in some special cases.

This fix is only intended for Linux, since the test is known to fail on Windows. But since we're currently only running Valgrind on Linux, this will give us improved control over memory issues.

BUG=
TEST=tools/valgrind-webrtc/webrtc_tests.sh -t cmdline out/Debug/video_processing_unittests

Review URL: http://webrtc-codereview.appspot.com/349012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 08:40:55 +00:00
pwestin@webrtc.org
56ee5d5d98 Bugfix 32 bit linux. 
Review URL: https://webrtc-codereview.appspot.com/353010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 07:47:38 +00:00
pwestin@webrtc.org
95cf47932d Remove list wrapper from FEC code. 
Review URL: https://webrtc-codereview.appspot.com/350013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 06:59:06 +00:00
leozwang@webrtc.org
9165f1fe91 Changed to use std::sort 
Review URL: https://webrtc-codereview.appspot.com/356003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-20 01:39:13 +00:00
leozwang@webrtc.org
a191506ce9 Enable all modules without building errors 
Review URL: https://webrtc-codereview.appspot.com/360004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 22:11:37 +00:00
marpan@webrtc.org
20cd06123c For TL(temporal layers) = 2, the alt-ref frame should not be used as a reference. 
Correction for the last frame in the cycle.
Review URL: https://webrtc-codereview.appspot.com/343015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 18:25:23 +00:00
pwestin@webrtc.org
0074187436 Removed map_wrapper from rtp_sender 
Review URL: https://webrtc-codereview.appspot.com/343014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:56:10 +00:00
pwestin@webrtc.org
3c9be1bc4d Removed list wrapper fromr overuse detector. 
Review URL: https://webrtc-codereview.appspot.com/353004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:55:54 +00:00
pwestin@webrtc.org
d4adc5b26f removed unused include from remote rate control. 
Review URL: https://webrtc-codereview.appspot.com/350015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:55:30 +00:00
pwestin@webrtc.org
af6f15c1bf Changed RTP reveivers to use stl map and list. 
Review URL: https://webrtc-codereview.appspot.com/349010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:59 +00:00
pwestin@webrtc.org
38f4816737 Removed unused include from rtp sender audio. 
Review URL: https://webrtc-codereview.appspot.com/348012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:33 +00:00
pwestin@webrtc.org
26f8d9c7f3 Removed list and map wrappers, for RTCP handling. 
Review URL: https://webrtc-codereview.appspot.com/349011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 15:53:09 +00:00
tina.legrand@webrtc.org
d71a11c15e Fixing Coverity issues in Audio Coding Module 
10198: Out-of-bounds read in acm_isac.cc
10251: Unintended sign extension in acm_resampler.cc
10273: Uninitialized pointer field in acm_amr.cc
10274: Uninitialized pointer field in acm_amrwb.cc
10275: Uninitialized scalar field in acm_dtmf_detection.cc
10276: Uninitialized pointer field in acm_g722.cc
10277: Uninitialized pointer field in acm_g7221.cc
10278: Uninitialized pointer field in acm_g7221c.cc
10279: Uninitialized pointer field in acm_g729.cc
10280: Uninitialized pointer field in acm_g7291.cc
10281: Uninitialized pointer scalar in acm_generic_codec.cc
10282: Uninitialized pointer field in acm_gsmfr.cc
10283: Uninitialized scalar field in acm_isac.cc
10284: Uninitialized pointer field in acm_opus.cc
10285: Uninitialized scalar field in acm_resampler.cc
10286: Uninitialized pointer field in acm_speex.cc
10287: Uninitialized scalar field in audio_coding_module_impl.cc
10581: Unintended sign extension in audio_coding_module_impl.cc

Additional change: removed unused function and member from ACMResampler.

Review URL: https://webrtc-codereview.appspot.com/343016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 13:22:22 +00:00
henrik.lundin@webrtc.org
dcdb744eee Remove an old comment in vp8 wrapper 
The operation that the comment describes was removed in r482.

Review URL: https://webrtc-codereview.appspot.com/353008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 13:06:43 +00:00
pwestin@webrtc.org
1da2327473 Changing header extension to use stl map. 
Review URL: https://webrtc-codereview.appspot.com/350014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 12:58:53 +00:00
stefan@webrtc.org
8e50693736 Fixes for code analysis warnings. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 12:30:21 +00:00
bjornv@webrtc.org
12cccddc63 NS-SWB: Actived SWB processing at once, i.e., no startup phase. 
Performance verified on a few 32 kHz files.
BUG=
TEST=audioproc, audioproc_unittest

Updated output_data_float.pb
Changes in SWB tests (3, 6, 9 and 12) as

Running test 3 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 1363
Expected: test->max_output_average()
Which is: 1386

Running test 6 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 2070
Expected: test->max_output_average()
Which is: 2109

Running test 9 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 1314
Expected: test->max_output_average()
Which is: 1336

Running test 12 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 2049
Expected: test->max_output_average()
Which is: 2085
Review URL: https://webrtc-codereview.appspot.com/344013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-19 08:56:38 +00:00
andrew@webrtc.org
267ca3162b Fix comparison-always-true warning with -Wextra. 
TEST=build on Linux with -Wextra.

Review URL: https://webrtc-codereview.appspot.com/353005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 19:41:40 +00:00
bjornv@webrtc.org
ab2bb82ac9 VAD refactor: int return value for Init. 
For consistency and as part of style, the return value of WebRtcVad_Init() has been changed to int.

Impact:
 1) audio_processing, audio_coding, a test in CNG, functionTest in audio_conference_mixer, a test in net_eq all used int values. Hence, unaffected.
 2) Function pointers in net_eq changed.
 3) The VADInit in neteq/dsp.c boiled down to typecast into int anyhow, which now is removed.

TEST=vad_unittests, neteq_unittests
Review URL: https://webrtc-codereview.appspot.com/355003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 14:51:02 +00:00
henrik.lundin@webrtc.org
4407edc314 Bugfix in VP8 packetizer 
Handle the case with no small partitions in Vp8PartitionAggregator.
Also added a new unit test for the packetizer to verify that the
bug is fixed.

TEST=RtpFormatVp8Test.TestAggregateModeTwoLargePartitions
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/348011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 10:01:03 +00:00
henrik.lundin@webrtc.org
7f2c2a5db2 Adding optimized aggrgation to VP8 packetizer 
This change introduces a new algorithm for aggregating small
partitions into packets. The algorithm is based on a tree-search
to find an optimal allocation of the packets, such that the
difference in size between packets is minimized.

The new method is used when partition aggregation is allowed and
balanced packets are requested. Otherwise, the old method is used.

The new method is implemented using the new classes
Vp8PartitionAggregator and PartitionTreeNode. Both classes have
dedicated unit tests.

In order to facilitate the new algorithm, the packetizer was
redesigned to calculate all packet sizes when the first packet is
extracted. The information about all packets is stored in a packet
queue structure, which is then popped for each packet extracted.

Finally, a bug in the old packetizer algorithm was fixed. The bug
caused a +/-1 error in packet sizes when balanced packets were
produced. The unit test were updated accordingly.

TEST=rtp_rtcp_unittests: PartitionTreeNode.* Vp8PartitionAggregator.* RtpFormatVp8Test.*

Review URL: https://webrtc-codereview.appspot.com/345008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-18 08:21:15 +00:00
andrew@webrtc.org
975e4a33c6 Fix gcc warnings triggered by -Wextra. 
TEST=build and audio_coding_unittests and audio_coding_module_test on Linux

Review URL: https://webrtc-codereview.appspot.com/343010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 19:27:33 +00:00
pwestin@webrtc.org
5621057956 Removing unused code. 
Review URL: https://webrtc-codereview.appspot.com/349008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 12:45:47 +00:00
pwestin@webrtc.org
df9bd9bdbd Removed dead code. 
Review URL: https://webrtc-codereview.appspot.com/352010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 11:42:02 +00:00
pwestin@webrtc.org
aafa5a331c Coverty report: Unititialized members 
Review URL: http://webrtc-codereview.appspot.com/349007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-17 07:07:37 +00:00
asapersson@webrtc.org
43b8fc5c0d Review URL: http://webrtc-codereview.appspot.com/345011 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 13:49:04 +00:00
stefan@webrtc.org
8ddf9a4e18 Ported more jitter buffer tests to unit tests. 
BUG=
TEST=jitter_buffer_unittest

Review URL: http://webrtc-codereview.appspot.com/350009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 11:59:01 +00:00
asapersson@webrtc.org
869ce2d441 Review URL: http://webrtc-codereview.appspot.com/353002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 11:58:36 +00:00
asapersson@webrtc.org
0b3c35a258 Review URL: http://webrtc-codereview.appspot.com/321011 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 11:06:31 +00:00
henrika@webrtc.org
f75901fa4c Resolves CID 10540: Copy into fixed size buffer (STRING_OVERFLOW). 
You might overrun the 32 byte fixed-size string "receiveCodec.plname" by copying "payloadName" without checking the length.
Note: This defect has an elevated risk because the source argument is a parameter of the current function.
Review URL: http://webrtc-codereview.appspot.com/352009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-16 08:45:42 +00:00
andrew@webrtc.org
c8d012fb32 Use -msse2 for SSE2 optimized code. 
When targeting 32-bit Linux, we need to pass -msse2 to gcc to compile
SSE2 intrinsics. However, -msse2 also gives gcc license to automatically
generate SSE2 instructions wherever it pleases. This will crash our code
on processors without SSE2 support.

This change breaks the files with SSE2 intrinsics into separate targets,
such that we can limit the scope of -msse2 to where it's needed.

We no longer need to employ the WEBRTC_USE_SSE2 define; the build system
decides when SSE2 is supported and compiles the appropriate files.

TBR=bjornv@webrtc.org
TEST=audioproc (performance testing), audioproc_unittest, video_processing_unittests, build on Linux (targeting ia32/x64, with disable_sse2==0/1), Mac, Windows

Review URL: http://webrtc-codereview.appspot.com/352008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 19:43:09 +00:00
andrew@webrtc.org
ee3fe5b982 Remove unused variable from mixer module. 
R=henrike@webrtc.org
BUG=coverity-10288

Review URL: http://webrtc-codereview.appspot.com/344010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 17:54:57 +00:00
braveyao@webrtc.org
5f9a7baaea Review URL: http://webrtc-codereview.appspot.com/347012 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 10:22:44 +00:00
mflodman@webrtc.org
117c119501 Only update REMB value if there is a calid bitrate estimate. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/352005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 08:52:58 +00:00
henrik.lundin@webrtc.org
33d5f69d5e Fix issue 218 with new solution 
This one is slightly more elegant and efficient.

BUG=http://code.google.com/p/webrtc/issues/detail?id=218
TEST=

Review URL: http://webrtc-codereview.appspot.com/344009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 07:46:50 +00:00
andrew@webrtc.org
7859e10985 Propagate decoding errors to the mixer module. 
Review URL: http://webrtc-codereview.appspot.com/348001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-13 00:30:11 +00:00
stefan@webrtc.org
c8277db7c8 Fix selective retransmissions after corrupt merge in r1373. 
BUG=228
TEST=

Review URL: http://webrtc-codereview.appspot.com/345006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 15:38:50 +00:00
pwestin@webrtc.org
9cbe6867e7 Removed experiment. 
Review URL: http://webrtc-codereview.appspot.com/345005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 15:35:28 +00:00
stefan@webrtc.org
ad4af57abd Fixes a jitter buffer NACK bug. 
If no frame has been decoded the jitter buffer might generate huge
erroneous NACK lists.

Adds a couple of new jitter buffer unittests (some ported from
jitter_buffer_test.cc).

Adds a test to the VCM robustness tests.

BUG=226
TEST=VCMRobustnessTest, TestJitterBufferFull, TestNackListFull, TestNackBeforeDecode, TestNormalOperation

Review URL: http://webrtc-codereview.appspot.com/352002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 15:16:49 +00:00
mflodman@webrtc.org
80d60420ff RTCPSender::_bitrate_observer not initialized. 
BUG=227
TEST=Valgrind

Review URL: http://webrtc-codereview.appspot.com/352004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 14:28:53 +00:00
henrik.lundin@webrtc.org
053c7991e3 Add minimum waiting time to NetEQ metrics 
Adding minWaitingTimeMs to ACMNetworkStatistics and to
NetworkStatistics. Also adding unittest.

TEST=audio_coding_unittests

Review URL: http://webrtc-codereview.appspot.com/350006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 14:16:44 +00:00
kjellander@webrtc.org
b39a3b4a7a Restoring unintentially renamed MS DirectShow source files in 
http://webrtc-codereview.appspot.com/348005/

BUG=
TEST=Compiling on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/352003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 12:22:03 +00:00
kjellander@webrtc.org
7f3c724e12 Renaming 47 files from .cpp to .cc 
In addition to our naming guidelines, this will cause these files to get parsed by Sonar, and to make searching/grepping the source using file extensions easier in the future.

BUG=
TEST=Compiling on Linux.

Review URL: http://webrtc-codereview.appspot.com/348005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 10:23:41 +00:00
kjellander@webrtc.org
93546f8ed6 Removing unused file 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/347010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 10:00:33 +00:00
niklas.enbom@webrtc.org
553657b2f8 See http://codereview.chromium.org/9188022/ for details 
Review URL: http://webrtc-codereview.appspot.com/347009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-12 08:49:34 +00:00
andrew@webrtc.org
83c7f6db0e Add missing file to iSAC gyp. 
TBR=kma@webrtc.org
TEST=Linux build

Review URL: http://webrtc-codereview.appspot.com/344008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 20:16:32 +00:00
andrew@webrtc.org
921321ff62 Fix unused-variable warning in iSAC. 
TBR=kma@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/348006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 19:50:20 +00:00
kma@webrtc.org
badf2b8044 Optimized an AR function in iSAC fix for ARMv7 (not Neon) platforms. 
Bit exact. Speed doubled.
Review URL: http://webrtc-codereview.appspot.com/327001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 18:01:39 +00:00
mflodman@webrtc.org
04c18cb37a Update all child modules of with received bandwidth estimate. 
BUG=224

Review URL: http://webrtc-codereview.appspot.com/347007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 14:18:33 +00:00
stefan@webrtc.org
cd8cea50a6 Fix decode error in NACK/FEC mode after network glitch. 
Caused when recyclingframes until the next key frame to
regain frame buffers when the jitter buffer is full.

BUG=http://code.google.com/p/webrtc/issues/detail?id=225
TEST=

Review URL: http://webrtc-codereview.appspot.com/350005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 14:17:44 +00:00
perkj@webrtc.org
ce5990cb0b Fix defect http://code.google.com/p/webrtc/issues/detail?id=222 
"ViE GetSentRTCPStatistics fails on a sending channel if it don't receive rtp video packets.

BUG=222
TEST= tested in loopback. No new test added yet.

Review URL: http://webrtc-codereview.appspot.com/343003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 13:00:08 +00:00
tina.legrand@webrtc.org
6b6ff558a8 Implementation if mono-to-stereo and vice versa in ACM. 
Added stereo-to-mono and mono-to-stereo tests to end of TestStereo.cpp.

BUG=Aim to resolve issue 207, "Investigate stereo capture handling in modules"
TEST=audio_coding_module_test

Review URL: http://webrtc-codereview.appspot.com/345002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 10:12:54 +00:00
braveyao@webrtc.org
e3eaf44ccf one logical enhancement in CoreAudio error handler. It should never happen, but so far the only suspect to a rare crash report. 
Review URL: http://webrtc-codereview.appspot.com/349002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-11 03:07:52 +00:00
stefan@webrtc.org
c5b73e3974 Further cleanup of OverUseDetector. Removed member no longer used. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 16:42:09 +00:00
pwestin@webrtc.org
a1783598a7 Bugfix for clang. 
Review URL: http://webrtc-codereview.appspot.com/351001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 15:33:40 +00:00
pwestin@webrtc.org
5d35ceb34a Bugfix array length in test. 
Review URL: http://webrtc-codereview.appspot.com/343007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 15:06:09 +00:00
pwestin@webrtc.org
8281e7dd4a Added RTX to ViE. 
Review URL: http://webrtc-codereview.appspot.com/336001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 14:09:18 +00:00
tina.legrand@webrtc.org
ac4eb046e3 Added registration of RED and CNG to NetEq slave. 
Bug found when working on issue 221. Missing registration of CNG was the cause of the bad audio (master and slave out of sync) reported in the issue.

NOTE! File has not been refactored to follow Google style.

BUG=http://code.google.com/p/webrtc/issues/detail?id=221

Review URL: http://webrtc-codereview.appspot.com/342006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 13:59:55 +00:00
henrik.lundin@webrtc.org
d4e8c0b3ff Fixing Issue 218 
Taking care of the case when the raw waiting times vector from
NetEQ is zero length.

Also adding a new unittest to cover this case.

BUG=http://code.google.com/p/webrtc/issues/detail?id=218
TEST=AcmNetEqTest.TestZeroLengthWaitingTimesVector

Review URL: http://webrtc-codereview.appspot.com/349003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 13:46:06 +00:00
asapersson@webrtc.org
c5a1cee73e Review URL: http://webrtc-codereview.appspot.com/348004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 12:54:44 +00:00
stefan@webrtc.org
727e1611ac Removes debug file writing. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/343006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 11:45:47 +00:00
stefan@webrtc.org
b07aa403b3 Fixes issue 210. Removes diff between two different arrays. 
Also fixes the FrameBuffer copy constructor.

BUG=210
TEST=

Review URL: http://webrtc-codereview.appspot.com/347002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 11:45:05 +00:00
stefan@webrtc.org
e21a8cf4d4 Fix issue 211: Make sure we always generate at least one FEC packet per frame if we need protection. 
BUG=211
TEST=

Review URL: http://webrtc-codereview.appspot.com/348002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-10 08:37:33 +00:00
marpan@webrtc.org
2dad3fbe49 Media-opt: Added a filter type mode for the filtering of the received packet loss. This makes the filter selection explicit and easier to modify/test. 
Removed the function UpdateLossPr(); the filter updates are done in the same function that returns the filtered loss.
Review URL: http://webrtc-codereview.appspot.com/333018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-09 18:18:36 +00:00
pwestin@webrtc.org
12d97f6637 Made send pad data generic (audio and video) 
Review URL: http://webrtc-codereview.appspot.com/343001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-05 10:54:44 +00:00
pwestin@webrtc.org
3aa25de346 Bugfix OnNetworkChanged not triggered for RTCP compund messages if TMMBR is higher than last value. 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/342001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-05 08:40:56 +00:00
wu@webrtc.org
d6b827a28e Fix for the build broken on Windows. 
Review URL: http://webrtc-codereview.appspot.com/335017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 22:38:05 +00:00
mikhal@webrtc.org
a58888d382 Updating capture module following latest libyuv api changes 
Review URL: http://webrtc-codereview.appspot.com/337009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 19:23:24 +00:00
mikhal@webrtc.org
7d5ca2be1f Updating render module following latest libyuv api changes. 
Review URL: http://webrtc-codereview.appspot.com/331019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 19:01:48 +00:00
kma@webrtc.org
746f9e31c0 Changed build settings for ARMv5 in Android. 
I found some issues in building ARMv5 with ICM. This CL includes fixes,
and a design change which now will exclude any NEON libraries unless 
the build is for dynamic detection or for Neon specifically.
Review URL: http://webrtc-codereview.appspot.com/330021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 17:47:57 +00:00
pwestin@webrtc.org
6c1d41583a Fix for RTP extension audio level. 
Review URL: http://webrtc-codereview.appspot.com/339002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 17:04:51 +00:00
andrew@webrtc.org
d77a6614fa Consts can't be used as C array size initializers. 
(Unless you happen to be using clang...)

TBR=bjornv@webrtc.org
TEST=build on gcc

Review URL: http://webrtc-codereview.appspot.com/333029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 16:22:24 +00:00
henrik.lundin@webrtc.org
d047b2e7f6 Enabling NetEQ unittest for more platforms 
Removing platform limitations for NetEqDecodingTest:TestBitExactness
and NetEqDecodingTest:TestNetworkStatistics. New reference files
where provided in revision 6 of the resources, which allows us
to enable these tests.

BUG=
TEST=neteq_unittests linux32/64, win32/64, mac32

Review URL: http://webrtc-codereview.appspot.com/329027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 16:10:23 +00:00
andrew@webrtc.org
3905b0c45d Protect against divide-by-zeros in AGC. 
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/333024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 15:47:20 +00:00
pwestin@webrtc.org
c450a19669 Removed Version function from all modules. 
TBR=henrik_a
Review URL: http://webrtc-codereview.appspot.com/329023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 15:00:12 +00:00
henrik.lundin@webrtc.org
d439870473 Adding two new network metrics to NetEQ 
Clock-drift (in parts-per-million) and peaky-jitter mode status.
Both metrics are propagated to the VoE API. Tests are added
in the NetEQ unittests, and to some extent in ACM unittests
and VoE tests.

Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.

Note: The file neteq_network_stats.dat in resources must be updated
for the unittests to pass.

Review URL: http://webrtc-codereview.appspot.com/337005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 13:09:55 +00:00
bjornv@webrtc.org
80d28b22b9 Changed to new ring buffer in AECM. 
Replaced the old ring buffer in AECM with the new one. Also removed the old one from ring_buffer.
Changes are bit exact according to audioproc_unittest fixed.

TEST=audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/331022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 09:55:09 +00:00