Commit Graph

1416 Commits

Author SHA1 Message Date
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
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
xians@webrtc.org
9b3474aff8 Disable the unused API interfaces for VoE chromium build.
Review URL: https://webrtc-codereview.appspot.com/377002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1592 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-02 08:59:11 +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
leozwang@webrtc.org
ac7e89ff1c Correct and update LICENSE
Review URL: https://webrtc-codereview.appspot.com/382001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1590 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-01 23:04:48 +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
bjornv@webrtc.org
26e8a58130 VAD refactor: Create() and Free()
Style and return value changes. No impact externally, since audio_processing, audio_conference_mixer and audio_coding either already assumes 'int' as return value, assumes nothing or doesn't take care of the return value.

TESTS=vad_unittests, audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/374006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1581 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-31 14:42:50 +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
xians@webrtc.org
79af734807 This patch fixes the converity warnings in voice engine.
Review URL: https://webrtc-codereview.appspot.com/373017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1579 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-31 12:22:14 +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
henrika@webrtc.org
2919e95c2a Resolves Coverty issue #10347.
Uninitialized member (UNINIT_CTOR).
Review URL: https://webrtc-codereview.appspot.com/369023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1577 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-31 08:45:03 +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
mflodman@webrtc.org
7627843352 Added NULL check in external transport test code.
BUG=C-10246

Review URL: https://webrtc-codereview.appspot.com/367022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1570 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-30 13:21:26 +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
phoglund@webrtc.org
048eb7cda6 Finished rewriting the audio processing test.
Partial rewrite of audio processing tests.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1561 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-27 11:58:41 +00:00
mflodman@webrtc.org
832adebca2 Removed MapWrapper from ViEFrameProviderBase.
BUG=C-10189

Review URL: https://webrtc-codereview.appspot.com/356002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1560 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-27 09:31:34 +00:00
mflodman@webrtc.org
194a93ac66 Adding ViE NULL checks.
BUG=C-10188, C-10246, C-10595

Review URL: https://webrtc-codereview.appspot.com/373013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1559 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-27 09:11:22 +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
mflodman@webrtc.org
9b0a820624 Fixed double erase in ViEChannelManager channel map.
Review URL: https://webrtc-codereview.appspot.com/369011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1546 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-25 16:00:49 +00:00
mflodman@webrtc.org
b11424bc11 Remove ViEShared inheritance for interface impl.
Review URL: https://webrtc-codereview.appspot.com/357002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1545 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-25 13:42:03 +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
bjornv@webrtc.org
2a4dcd7d15 VAD refactor: WebRtcVad_InitCore().
Impact only locally.
- Replaced for loops with memset().
- Added guard against NULL pointer.
- Removed mode as input parameter (never really used).
- Updated unit tests.
- Made struct member init_flag "int".
- Updated function description.
- Updated Copyright notes with 2012.
- Removed some lint warnings.

TESTS=vad_unittests, audioproc_unitest
Review URL: https://webrtc-codereview.appspot.com/369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1543 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-25 12:18:12 +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
perkj@webrtc.org
40d3c08be4 Changed max number of vie channels to 32.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/374002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1527 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 14:04:17 +00:00
mflodman@webrtc.org
ba09cf16ec Correcting uninitialized members.
BUG=C-10344, C-10345, C-10346

Review URL: https://webrtc-codereview.appspot.com/345012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1525 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 07:49:33 +00:00
mflodman@webrtc.org
a5a5cbb992 Switched from WebRTC wrappers to stl in ChannelManager.
BUG=C-10187

Review URL: https://webrtc-codereview.appspot.com/357001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1524 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 06:50:15 +00:00
andrew@webrtc.org
eeaf3d1fc1 Merge /branches/3.2:r1380 to /trunk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1523 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 06:30:02 +00:00
mflodman@webrtc.org
6cf529d082 Changed REMB return value to int instead of bool.
Review URL: https://webrtc-codereview.appspot.com/366001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1522 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 06:16:16 +00:00
andrew@webrtc.org
d3a0c1cb66 Merge /branches/3.2:r1378 to /trunk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1521 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 03:20:53 +00:00
kma@webrtc.org
4bc24c4afa Optimized function WebRtcSpl_FilterARFastQ12 for ARM platform.
Speed close to doubled for an offline test in NetEq.
Bit exact.
Review URL: https://webrtc-codereview.appspot.com/346001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1520 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-24 02:12:49 +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
f5cacdce8c Fix line aligement
Review URL: https://webrtc-codereview.appspot.com/373002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1516 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-23 23:14:13 +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
phoglund@webrtc.org
12dbc23851 Rewrote volume test.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/367004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1506 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-23 16:03:04 +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
phoglund@webrtc.org
3b57ee0238 Rewrote DTMF test.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/368001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1502 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-23 09:22:33 +00:00
leozwang@webrtc.org
31627fe82c Add vie_remb.cc to makefile
Review URL: https://webrtc-codereview.appspot.com/358004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1501 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 23:10:26 +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
leozwang@webrtc.org
2638577f03 Add an argument in ANDROID_NOT_SUPPORT macro
Review URL: https://webrtc-codereview.appspot.com/363003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1499 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 18:45:45 +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
phoglund@webrtc.org
d056abd62f Prepared tests for running on build bot.
The changes will mostly clarify why PSNR and SSIM thresholds are chosen. I think those explanations will be good since these tests may well be a bit flaky.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/343017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1496 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 12:48:13 +00:00
mflodman@webrtc.org
c672d34ac7 Removed file duplicate added in r1312.
Review URL: https://webrtc-codereview.appspot.com/357003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1495 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 12:30:42 +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
andrew@webrtc.org
f33dfa89b9 Add target and config info to merged lib name.
BUG=None
TEST=build merged_lib on Linux, Mac, Win

Review URL: https://webrtc-codereview.appspot.com/344014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1487 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-20 01:32:16 +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
mflodman@webrtc.org
2d03b8b217 REMB now works for two consecutive calls with different channels but same ViE instance.
BUG=241

Review URL: https://webrtc-codereview.appspot.com/361001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1484 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 21:21:15 +00:00
andrew@webrtc.org
300aed945d Use -Wextra on Linux for standalone builds.
BUG=None
TEST=build on Linux, Mac, Windows

Review URL: https://webrtc-codereview.appspot.com/348009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1482 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 20:10:19 +00:00
mflodman@webrtc.org
0f4cb131c6 Added NULL check in ViEFileImpl.
BUG=C-10188

Review URL: https://webrtc-codereview.appspot.com/344016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1480 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 18:41:39 +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
tommi@webrtc.org
9ff87db5c0 Remove the diamond inheritance pattern from VoEVideoSyncImpl in attempt to see if this fixes coverity reports.
CID=10446,10445,10444,10443
Review URL: https://webrtc-codereview.appspot.com/343018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1472 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 15:05:36 +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
mflodman@webrtc.org
70111e69bf Switching to stl instead of WebRTC wrappers.
BUG=C-10436, C-10437, C-10438

Review URL: https://webrtc-codereview.appspot.com/343012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1468 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 12:48:53 +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
6748087467 Incorrect pointer inputs in filter test.
Two filter functions were called wrongly, which made us read from uninitialized values.
Added short comment.
No style changes. The tests will be revisited and extended during an upcoming refactor.

signal_processing_unittests now pass Valgrind.

TEST=signal_processing_unittests
Review URL: https://webrtc-codereview.appspot.com/344015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1466 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 10:41:39 +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
17585856f5 Merged /branches/3.2:r1381 to /trunk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1464 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-19 00:15:08 +00:00
mflodman@webrtc.org
d5a83ce972 Changed to red and ULPFEC to ulpfec in GetCodec.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/355005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1462 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 22:39:32 +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
punyabrata@webrtc.org
ad1927d368 Changing the typing detection sensitivity as the current
setting does not work well in some scenarios especially
using webcams with built-in microphones.
Review URL: https://webrtc-codereview.appspot.com/349009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1455 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 18:53:04 +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
phoglund@webrtc.org
5badc7e969 Put system cpu tests back in, improved documentation.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/350011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1452 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 10:46:07 +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
mflodman@webrtc.org
8224451ee4 Add check for ftell return value.
BUG=C-10170

Review URL: https://webrtc-codereview.appspot.com/355001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1450 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 09:36:46 +00:00
mflodman@webrtc.org
cdeb483c6a Fixed ignored return value.
BUG=C-10011

Review URL: https://webrtc-codereview.appspot.com/353003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1449 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 09:00:04 +00:00
mflodman@webrtc.org
a768ca13f4 Removed dead code.
BUG=C-10062, C-10063, C-10064, C-10065, C-10393, C-10394.

Review URL: https://webrtc-codereview.appspot.com/343013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1448 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-18 08:52:16 +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
bjornv@webrtc.org
4259fd725c Refactor VAD: Code restructure
- Tests added for vad_core.
- Replaced two macros with constants.
- Made an internal function static.
- Replaced replicated code with function call.
Review URL: https://webrtc-codereview.appspot.com/354001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1444 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 14:37:59 +00:00
pwestin@webrtc.org
38e0a771d2 Bugfix removed MPEG4 from windows test.
Review URL: https://webrtc-codereview.appspot.com/348010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1443 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 14:18:09 +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
mflodman@webrtc.org
e5297d2aaa Big parameter passed as argument.
BUG=C-10503, C-10504, C-10505

Review URL: https://webrtc-codereview.appspot.com/343011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1441 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 12:44:41 +00:00
mflodman@webrtc.org
2877bdc590 Cleaned up resource leaks.
BUG=C-10059, C-10228, C-10229.

Review URL: https://webrtc-codereview.appspot.com/345013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1439 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 12:18:16 +00:00
mflodman@webrtc.org
7b3f3b1e42 CalcBufferSize can return -1, which wasn't handled by ViERenderer.
BUG=C-10532

Review URL: https://webrtc-codereview.appspot.com/345010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1438 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-17 12:12:42 +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
phoglund@webrtc.org
c12f815de6 Rewrote hardware test and fixed broken tests on Windows.
Fixed broken tests on Windows, including old tests.

Rewrote hardware test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/347008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1434 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 12:40:18 +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
mflodman@webrtc.org
67cdc22e7e CpuLinux file handle leak.
BUG=crbug.com/110165

Review URL: http://webrtc-codereview.appspot.com/353001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1429 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 09:31:39 +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
braveyao@webrtc.org
f5c6573725 fix defect http://code.google.com/p/webrtc/issues/detail?id=215, audio device is not stopped appropriately.
Review URL: http://webrtc-codereview.appspot.com/350008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1427 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-16 03:04:46 +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
perkj@webrtc.org
5735a63e5a Add video capture module to the list of dependent projects in video engine.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/348007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1409 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 14:22:58 +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
bjornv@webrtc.org
40ea5106f6 Refactoring vad_filterbank
Made internal function LogOfEnergy() more efficient.
Includes
- Name change "vector" -> "data"
- Complete refactor of LogOfEnergy()
- Removed lint warning

Major changes:
* Removed unnecessary variables
* Reduced number of shifts
* Removed one norm calculation


TEST=vad_unittests, audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/347004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1407 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 12:47:42 +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
punyabrata@webrtc.org
9a85b500c5 Minor tracing fix in ::IncomfingFrame and ::IncomfingFrameI420
Review URL: http://webrtc-codereview.appspot.com/352001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1401 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-12 00:37:07 +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
mflodman@webrtc.org
684c7b71c3 Fixed vie_defines.h typo.
BUG=220
TEST=Mac debug build

Review URL: http://webrtc-codereview.appspot.com/347006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1389 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 13:52:06 +00:00
mflodman@webrtc.org
5007056767 Added REMB option to custom call.
Review URL: http://webrtc-codereview.appspot.com/347003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1388 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 13:46:10 +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
phoglund@webrtc.org
01530a2ac2 Rewrote the rcp_rtcp test.
Finished rewriting the rtp_rtcp test.

Rewrote first RTP RTCP test

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/342007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1386 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 12:26:34 +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
pwestin@webrtc.org
df9866fedb Bugfix mac pid_t
Review URL: http://webrtc-codereview.appspot.com/350004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1384 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 08:57:47 +00:00
pwestin@webrtc.org
b54d72778c Changed thread Id handling in trace.
Review URL: http://webrtc-codereview.appspot.com/331020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1383 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-11 08:28:04 +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
bjornv@webrtc.org
d1f148da77 Refactor vad_filterbank: Some restructuring.
- Removed unnecessary type casting.
- Added comments.
- Removed shift macros.
- Name change of _get_features() to _CalculateFeatures(). Affects vad_core.c and vad_filterbank_unittest.cc.
Review URL: http://webrtc-codereview.appspot.com/343002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1371 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 13:48:09 +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
phoglund@webrtc.org
caf39f335f Re-enabled RTP-RTCP test since it's not flaky anymore.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/345003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1369 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-10 13:43:33 +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
mflodman@webrtc.org
0ab8ba313b We now require a manually set sender to send REMB packets.
BUG=
TEST=video_engine_unittests

Review URL: http://webrtc-codereview.appspot.com/348003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1358 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 16:16:49 +00:00
bjornv@webrtc.org
d9c87b2146 Refactor vad_filterbank: Local functions made static.
Review URL: http://webrtc-codereview.appspot.com/342002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1357 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 13:11:29 +00:00
phoglund@webrtc.org
d8d85711c7 Temporarily disabled the standard rtp-rtcp test because of flakiness.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/349001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1356 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 12:06:42 +00:00
phoglund@webrtc.org
0aa7b32652 Finished rewriting the codec test.
Rewrote more tests.

Rewrote most of the codec test and removed it from the regular test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1355 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 11:15:46 +00:00
phoglund@webrtc.org
dc9536dd0e Made vie_auto_test more robust in Linux when the X environment is broken.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1354 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 10:49:23 +00:00
mflodman@webrtc.org
0c0216f3f6 Correcting typo in libyuv.h.
Review URL: http://webrtc-codereview.appspot.com/333026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1353 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 10:27:46 +00:00
bjornv@webrtc.org
e6471ba8d2 VAD unittest updates.
Split the local function tests into separate files.
Review URL: http://webrtc-codereview.appspot.com/330031

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1352 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-09 09:54:07 +00:00
kma@webrtc.org
b0abbd353d Optimized spl function WebRtcSpl_CrossCorrelation for ARM Neon platforms.
When used in Neteq, Neteq performance improved from 13 to 33% with different
test configurations.
Output is not bit-exact with generic C code in file cross_correlation.c, 
due to reduction of shift operations from using Neon registers, although in
theory now the result is more accurate than before.
Review URL: http://webrtc-codereview.appspot.com/333013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1350 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-06 19:50:20 +00:00
leozwang@webrtc.org
bccac66885 Use a more common macro to get thread id
Review URL: http://webrtc-codereview.appspot.com/342004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1349 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-06 00:32:00 +00:00
mikhal@webrtc.org
a2026ba4c4 libyuv: Removing old unused functionality
Review URL: http://webrtc-codereview.appspot.com/329020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1347 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-05 18:19:32 +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
bjornv@webrtc.org
8f4a4ce13b Refactoring vad_filterbank: Style changes.
Consists of:
- variable names.
- variable initialization.
- ordered input/output parameters.

TEST=vad_unittest, audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1345 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-05 08:42:39 +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
punyabrata@webrtc.org
a0211c38ca Updating video revision
Review URL: http://webrtc-codereview.appspot.com/335016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1339 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 20:02:00 +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
mikhal@webrtc.org
d61e1cab08 Updating video engine following latest libyuv api changes
Review URL: http://webrtc-codereview.appspot.com/330026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1336 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 18:57:52 +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
kjellander@webrtc.org
94558d83bf Fixing Valgrind warnings caused by open files and undeleted memory.
Restructured scaler_unittest.cc to focus tests on testing one thing.

BUG=
TEST=libyuv_unittests in Debug+Release at Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/329026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1329 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 13:51:50 +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
bjornv@webrtc.org
226c5a1a95 Refactoring of vad_sp.[h/c]
- define guard name change
- changed to stdint
- added unit test
- removed shift macros
- style changes
- comments
Review URL: http://webrtc-codereview.appspot.com/336004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1326 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 09:15:12 +00:00
kjellander@webrtc.org
cc33737a80 Changing all PSNR/SSIM calculations to use libyuv.
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.

BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/333025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 08:09:32 +00:00
turaj@webrtc.org
a574b1c617 The inline implementation of WebRtcIsac_lrint(), which was implemented in several files, is now os_specific_inline.h. Define guards are modified according to WebRtc OS macros.
This resolves BUG=issue137.
Review URL: http://webrtc-codereview.appspot.com/269014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1323 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-04 02:26:23 +00:00
mikhal@webrtc.org
cd64886a2f video_coding: Updating NACK functions naming
Review URL: http://webrtc-codereview.appspot.com/329018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1322 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 23:59:42 +00:00
punyabrata@webrtc.org
8fa31bc4e5 Truncated messages, need a %S instead of $s for a double byte TCHAR
Review URL: http://webrtc-codereview.appspot.com/333002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1321 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 22:34:15 +00:00
mflodman@webrtc.org
adec9271b0 Correcting VieChannelManager bug.
Review URL: http://webrtc-codereview.appspot.com/337010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1320 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 21:43:15 +00:00
amyfong@webrtc.org
de5a10a044 Added in setting the minimum bit rate of a codec to ViE Custom Call
Review URL: http://webrtc-codereview.appspot.com/333019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1319 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 21:06:26 +00:00
mikhal@webrtc.org
77c425b976 video_coding: Checking/updating seq num for an old packet regardless of size.
Review URL: http://webrtc-codereview.appspot.com/330028

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1318 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 20:35:25 +00:00
mikhal@webrtc.org
c00f91d62d Adding BGRA as a video type.
This CL is a prerequisite for the capture module update CL. 
Review URL: http://webrtc-codereview.appspot.com/329021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1317 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 18:49:15 +00:00
andrew@webrtc.org
877c54e230 Fix unused-variable warning in Release.
TBR=mflodman@webrtc.org
TEST=Build Debug/Release on Linux

Review URL: http://webrtc-codereview.appspot.com/338003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1316 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 18:32:01 +00:00
bjornv@webrtc.org
f175125e96 Refactoring vad_filterbank: Style changes.
Includes:
- Correct header guard
- Indentations and white spaces
- Changed to stdint
Review URL: http://webrtc-codereview.appspot.com/330030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1315 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 15:07:42 +00:00
mflodman@webrtc.org
9c0aedc28b Removed constraint for changing resolution when using default encoder and added VP8 log.
Review URL: http://webrtc-codereview.appspot.com/330029

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1314 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 13:46:49 +00:00
henrik.lundin@webrtc.org
6c877363f7 Fix formatting for some NetEQ test tools
Format and lint for RTPchange.cc, RTPcat.cc and RTPanalyze.cc.

Review URL: http://webrtc-codereview.appspot.com/329024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1313 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 10:03:19 +00:00
perkj@webrtc.org
60c9bbd976 Fix GetReceivedRTCPStatistics and GetSendRTCPStatistics.
Comments where wrong and removed error message when trying to get RTT time from GetReceivedRTCPStatistics.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/335013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1312 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-03 09:54:29 +00:00
mflodman@webrtc.org
d5a4d9bce6 First refactoring of ViE interface.
Review URL: http://webrtc-codereview.appspot.com/337003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1311 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-02 13:04:05 +00:00
kjellander@webrtc.org
a643d5c4ef Integration test for videoprocessor
Added temporal layers number flag for video_quality_measurement tool.
This tool now also uses webrtc::VideoCodingModule::Codec() to get its
VideoCodec struct configuration instead of filling it in manually.

Updated paths for header files to use full directory paths.

Tested in Debug+Release on Linux, Mac and Windows. Passes Valgrind memcheck on Linux.

BUG=
TEST=video_codecs_test_framework_integrationtests. Also executed out/Debug/video_quality_measurement --input_filename=resources/foreman_cif.yuv  --width=352 --height=288

Review URL: http://webrtc-codereview.appspot.com/339001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1310 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-30 14:44:07 +00:00
mikhal@webrtc.org
62665b8cd3 video_coding: Adding a unit test to the decodableState class
Review URL: http://webrtc-codereview.appspot.com/315001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1309 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 18:09:58 +00:00
mikhal@webrtc.org
9eeafbef3c Updating the frame buffer return value in InsertPacket: Return NoError when a packet is inserted to a frame which is being decoded.
Review URL: http://webrtc-codereview.appspot.com/330027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1308 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:38:56 +00:00
mikhal@webrtc.org
bed34a341a video_coding: Updating seq number for old zero size packets. Updating function name to reflect zero size packets and not empty packets.
Review URL: http://webrtc-codereview.appspot.com/333009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1307 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 17:37:39 +00:00
bjornv@webrtc.org
70adcd46b2 Delay estimator improvements.
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.

TEST=audioproc_unittest + offline file tests.

output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 14:51:21 +00:00
stefan@webrtc.org
efd0a48c61 Add error resilient mode options to the VP8 specific VideoCodec struct.
It is useful to disable error resilience when we know we won't decode
with errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1305 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-29 10:12:35 +00:00
mikhal@webrtc.org
67f294a48a Adding a return value to ConvertRotationMode
Review URL: http://webrtc-codereview.appspot.com/333023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1304 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 23:28:45 +00:00
andrew@webrtc.org
6d6a43d6e3 Use char as ring-buffer data type.
- Avoids a bunch of char* casts.
- Use enum type rather than char.

TEST=audioproc_unittest on Linux (float and fixed), build on Windows

Review URL: http://webrtc-codereview.appspot.com/336010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1303 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 22:40:15 +00:00
mikhal@webrtc.org
e2642494e4 libyuv: Updating API to use latest ConvertFrom/To functionality
Review URL: http://webrtc-codereview.appspot.com/333020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1302 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 21:21:40 +00:00
bjornv@webrtc.org
267d0133ff Fixed pointer operations on void.
This should fix the error on Win where pointer arithmetics are done on void pointers. Type cast to char to interpret a size.
Review URL: http://webrtc-codereview.appspot.com/329019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1300 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 10:26:17 +00:00
bjornv@webrtc.org
7270a6bcc2 Merged apm-buffer branch [r1293] back to trunk.
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.

TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-28 08:44:17 +00:00
mikhal@webrtc.org
e39de16fa5 Moving video type convert functionality to libyuv. deleting vplibConversions as it is no longer needed.
Review URL: http://webrtc-codereview.appspot.com/338002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1298 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-27 23:45:30 +00:00
stefan@webrtc.org
f6c6b1c5b5 Include the media packet FEC headers in the video bitrate.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/328014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1296 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 10:33:39 +00:00
stefan@webrtc.org
39670f6aa6 Only reset the last decoded sequence number after flushing until key frame.
We can't reset the complete last decoded state when we recycle until a
key frame because that will allow any delta frame to be decoded afterwards,
and since the decoder isn't reset we will get decode errors.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1295 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-23 09:08:51 +00:00
mflodman@webrtc.org
1ce66e4dfb Don't report error when failing to send RTCP BYE.
Review URL: http://webrtc-codereview.appspot.com/337002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1293 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 18:40:15 +00:00
amyfong@webrtc.org
ee2924cc56 Added vp8 codec temporal layer changing option to ViE AutoTest custom call.
Review URL: http://webrtc-codereview.appspot.com/330018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1292 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 18:24:28 +00:00
mflodman@webrtc.org
d32c44738a Changed constructor used for CriticalSectionScoped in ViE.
Only changed:
- Name of some of the critsects.
- All critsects (but one) are now scoped_ptr.
- Use of ptr constructor of CriticalSectionScoped instead of reference version.

BUG=184
TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/330015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1291 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 14:17:53 +00:00
stefan@webrtc.org
6a4bef4e65 Implements selective retransmissions.
Default is set to not retransmit VP8 non-base layer packets or FEC packets.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1290 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:52:41 +00:00
mflodman@webrtc.org
51faeed6be Fixed REMB unit test on Windows.
TBR=pwestin

Review URL: http://webrtc-codereview.appspot.com/330022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1289 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:35:23 +00:00
pwestin@webrtc.org
f4d3b9d5a1 Cleaned up leaky symbols in NS.
Review URL: http://webrtc-codereview.appspot.com/337001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1288 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:33:08 +00:00
pwestin@webrtc.org
ebcb6421b1 Cleaned up leaky symbols in G722.
Review URL: http://webrtc-codereview.appspot.com/333017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1287 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:20:06 +00:00
pwestin@webrtc.org
d8f8b32521 Cleaned up leaky symbols in iSAC.
Review URL: http://webrtc-codereview.appspot.com/329014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1286 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 12:19:43 +00:00
stefan@webrtc.org
2ae4c8cf44 Disable temporal toggling by default.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1285 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:30:42 +00:00
mflodman@webrtc.org
84dc3d134d Add REMB functionality to ViE.
This CL only adds support for encoding one stream, but receiving multiple streams.

BUG=
TEST=video_engine_core_unittest + auto_test/loopback

Review URL: http://webrtc-codereview.appspot.com/333011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 10:26:13 +00:00
pwestin@webrtc.org
093ffad26b Removed unused function messing up the symbols.
Review URL: http://webrtc-codereview.appspot.com/336006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1283 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 09:48:48 +00:00
pwestin@webrtc.org
43761beb47 Bugfix get thread ID for linux.
Review URL: http://webrtc-codereview.appspot.com/331015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1282 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 09:40:28 +00:00
mflodman@webrtc.org
a4863dbdf0 Moved video_engine/main/interface to video_engine/include.
Only changed include paths in files, gyp-files and Android.mk.

TEST=vie_auto_test and peerconnection_client builds.

Review URL: http://webrtc-codereview.appspot.com/330017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1281 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:51:52 +00:00
henrik.lundin@webrtc.org
1e28d3c2e1 Change VP8 packetizer to use a single max payload size
The packetizer class is changed so that the max payload size is
provided on construction of the class rather than for each packet.
The tests are re-written to comply with the new design.

Also fixing a few errors in the tests.

Review URL: http://webrtc-codereview.appspot.com/335010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1280 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:49:31 +00:00
stefan@webrtc.org
f5edb923b1 Remove unused variable.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1279 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:34:31 +00:00
tina.legrand@webrtc.org
5c43b1b861 Updated resampler unit test with stereo.
I needed to run valgrind on this particular test, to exclude from valgrind warnings in ACM. Test passes valgrind without problems.
Review URL: http://webrtc-codereview.appspot.com/332010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1278 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 08:28:05 +00:00
pwestin@webrtc.org
8edb39db30 Prevent sending empty RTCP packet.
Review URL: http://webrtc-codereview.appspot.com/331009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1277 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 07:40:33 +00:00
henrik.lundin@webrtc.org
4a19030131 New VCM robustness API
This CL defines and starts to implement a new robustness API for
video coding module. The API is partly implemented. Some of the
modes and methods are still TBD.

Also including a new unittest with mocking of decoder and callbacks,
and faking of system clock.

Review URL: http://webrtc-codereview.appspot.com/333006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1276 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 20:38:37 +00:00
andrew@webrtc.org
697bc43b67 Restore item deletions in Windows UDP.
TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/331013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1275 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 19:58:11 +00:00
andrew@webrtc.org
71571c5446 Remove unneeded variables from windows UDP.
TEST=build on Windows.

Review URL: http://webrtc-codereview.appspot.com/329013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1274 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 18:30:59 +00:00
andrew@webrtc.org
3192d655bd Fix for devices lacking stereo support.
The number of capture channels can only be determined upon receiving the
first captured frame. We now assume stereo capture by default and set the
number of AudioProcessing input channels based on captured frames.

TEST=Windows mono-only device now runs AudioProcessing correctly (NS etc.), voe_auto_test (though some new, seemingly unrelated, tests are failing)

Review URL: http://webrtc-codereview.appspot.com/330013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1273 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 18:00:59 +00:00
andrew@webrtc.org
003044a6df Enable warnings-as-errors on Mac.
TEST=build on Mac (make and XCode)

Review URL: http://webrtc-codereview.appspot.com/335007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1272 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 16:36:56 +00:00
kjellander@webrtc.org
173b7bbc16 Integration test that tracks dropped frames and compares video output.
The recorded frame timestamps are used to modify the output video on a frame-per-frame so it can be compared with the reference video using PSNR. This code will make it possible to use vie_auto_test for full stack comparisons with network interference and similar interesting simulations.

There's some refactoring done in vie_comparison_test.cc to make it fit to the new test.

Compiled and executed in Debug+Release on Linux, Mac and Windows.

BUG=
TEST=vie_auto_test --automated --gtest_filter=ViEVideoVerificationTest.*

Review URL: http://webrtc-codereview.appspot.com/320002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1269 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 16:11:25 +00:00
mallinath@webrtc.org
03532b5f41 Fixing the double delete problem in UdpSocket2ManagerWindow. PopFront deletes the items, to there is no need to delete item explicitly.
Review URL: http://webrtc-codereview.appspot.com/333014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1268 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 15:36:44 +00:00
henrik.lundin@webrtc.org
7d8c72e2db Re-implement dependency injection of TickTime into VCM and tests
This change basicly re-enables the change of r1220, which was
reverted in r1235 due to Clang issues.

The difference from r1220 is that the TickTimeInterface was
renamed to TickTimeClass, and no longer inherits from TickTime.

Review URL: http://webrtc-codereview.appspot.com/335006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1267 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-21 15:24:01 +00:00