Commit Graph

1268 Commits

Author SHA1 Message Date
stefan@webrtc.org
a475556f5a Assume 200 ms RTT if we're only receiving.
BUG=
TEST=

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1728 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 09:00:35 +00:00
phoglund@webrtc.org
b45ceed9ef Rewrote the call report test.
BUG=
TEST=

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

BUG=
TEST=

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1712 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 00:38:33 +00:00
mallinath@webrtc.org
ee628358f4 Updating the object-c++ file after change in the API
GetBestMatchedCapability
Review URL: https://webrtc-codereview.appspot.com/396009

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

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

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

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

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

TEST=video_engine_core_unittests

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1705 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 14:50:24 +00:00
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
andrew@webrtc.org
cb333530fc Remove common_settings.gypi.
Now fully replaced by src/build/common.gypi.

BUG=
TEST=build

Review URL: https://webrtc-codereview.appspot.com/395003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1699 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 01:16:28 +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
leozwang@webrtc.org
0a272eb44b Disable SetAffinity on android
CPU_ macros are only available in android source tree, not in NDK. Disable it for now. 
Review URL: https://webrtc-codereview.appspot.com/392008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1697 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 22:35:29 +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
kjellander@webrtc.org
cd46385142 Fixing Android.mk for jpeg library
TBR=leozwang
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1692 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 10:12:52 +00:00
kjellander@webrtc.org
0a57aae75b Converted old jpeg_test tool to gtest unit test.
Restructured paths to new directory layout.

Stefan: common_video/*
Magnus: video_engine/*
Niklas: Android.mk

BUG=
TEST=jpeg_unittests on Debug+Release on Linux, Mac, Windows. Valgrind on Linux passes without warnings.

Review URL: https://webrtc-codereview.appspot.com/388007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1691 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 09:47:55 +00:00
andrew@webrtc.org
8bd6f19abe Disable flaky CpuTest.Usage on Windows.
TBR=turaj@webrtc.org
BUG=290
TEST=system_wrapper_unittests

Review URL: https://webrtc-codereview.appspot.com/396005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1689 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 01:47:51 +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
phoglund@webrtc.org
292da24166 New attempt.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1672 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 15:21:33 +00:00
phoglund@webrtc.org
dbe1e13b53 Fixed compilation error on Windows.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1670 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 14:03:44 +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
phoglund@webrtc.org
6b3bb89f12 Rewrote file test.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1668 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 12:14:54 +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
phoglund@webrtc.org
aaa76f3ba8 Rewrote network test.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/383003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1656 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 16:41:30 +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
454a27c13d The pthread_t is non-pointer type.
TBR=henrike
Review URL: https://webrtc-codereview.appspot.com/392004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1646 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 20:36:23 +00:00
henrike@webrtc.org
143abd95a3 Fixes coverity warnings in system_wrappers.
Review URL: https://webrtc-codereview.appspot.com/389003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1645 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 19:39:38 +00:00
henrike@webrtc.org
0e7c060256 Linux logs were not displaying time at ms resolution.
Review URL: https://webrtc-codereview.appspot.com/267012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1644 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 18:53:50 +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
xians@webrtc.org
594ab3ce4b remove vie file API to take away media_file and utility modules.
This CL reduce the size of chrome in release build by 70KB.
With this patch and r1592 , sizes.py reports 92255640 bytes with webrtc, down from 92485792 bytes.
The size is 88839360 bytes without webrtc.

BR,
/SX
Review URL: https://webrtc-codereview.appspot.com/380007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1637 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 10:38:12 +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
kma@webrtc.org
551fcc04ec Optimized function WebRtcSpl_DownsampleFast for ARM-NEON platform.
Review URL: https://webrtc-codereview.appspot.com/371001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1629 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 18:03:11 +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
kma@webrtc.org
59f16ec993 Introduced ARM version of WebRtcSpl_SqrtFloor(). Function cycles reduced by ~ 30% in a real time VOE test in an android device (Nexus-S, ARMv7a).
// Fritz, I added you as a reviewer for the assembly files, just as a warm-up for future storms. :-) The assembly code was from public domain and there's little to touch.
Review URL: https://webrtc-codereview.appspot.com/369017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1627 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 17:15:15 +00:00
phoglund@webrtc.org
9d9ad88ba5 Fixed remaining warnings.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/393001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1626 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 16:16:52 +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
bjornv@webrtc.org
530963925e Solves buffer overrun crash on Windows [issue 258].
Removed function calls not tested. Added a TODO on activating them when refactoring signal_processing.
Review URL: https://webrtc-codereview.appspot.com/379012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1620 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 08:10:46 +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
henrike@webrtc.org
2660460b89 Fixes flakyness in CPU unittest
Review URL: https://webrtc-codereview.appspot.com/377005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1616 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-06 23:33:54 +00:00
wu@webrtc.org
06c7dbae14 Disable flaky test AudioProcessingTest.TestVoiceActivityDetectionWithObserver.
BUG=263
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/380009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1615 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-06 23:13:21 +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
phoglund@webrtc.org
56b85c6ba8 Reduced potential for flakiness in voice detection tests.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1612 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-06 18:48:33 +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
kjellander@webrtc.org
ce0a6ff43d Restoring previous vie_auto_test.gypi structure due to problems on Mac
Now the unit test is included in the vie_auto_test target and executed when the automated flag is used.

TBR=mflodman
BUG=
TEST=vie_auto_test --automated --gtest_filter=FrameDropPrimitivesTest.FixOutputFileForComparison

Review URL: https://webrtc-codereview.appspot.com/381003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1598 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-03 14:06:46 +00:00
kjellander@webrtc.org
918a8bf40c External transport is modified to never drop packets from the first frame.
Refactoring of FrameDropHandler: It now also tracks when frames are leaving the encoder and is being sent to external transport.

Previous 'Sent' state is now renamed to 'Created'.

NOTICE: The test seems to be a little flaky on Linux so it's not ready for buildbots yet. Since this might be caused by unstable production code further investigation should be performed to clear out the flakiness. I will file an issue for this when this CL is submitted (since I don't have any code to refer to before that). Usually the flakiness is caused by a decoded/rendered callback that is left out for the last frame, but I have seen other flaky failures too, which means it's not as simple as ignoring the last frame.
These errors occur even if 400kbps bit rate and 0% PL and 0 delay is configured.

BUG=
TEST=vie_auto_test --automated --gtest_filter="ViEVideoVerificationTest.RunsFullStackWithoutErrors" in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1597 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-03 12:40:28 +00:00