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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
a8084b07e3 Revert r1628 which causes the crash of voe_auto_test. 
With r1628, it's possible the second memcpy got a NULL nearendClean.

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-08 12:42:56 +00:00
stefan@webrtc.org
23307f7c4b Remove frame_list.cc from Andorid.mk. 
TBR=mflodman
BUG=
TEST=

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

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

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

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

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

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

BUG=
TEST=

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

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

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

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

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

BUG=
TEST=audioproc_unittest

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

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

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

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

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

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

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

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

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

BUG=114
TEST=audioproc_unittest

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-02-07 00:01:04 +00:00
wu@webrtc.org
50099af75f Disable flaky test VideoProcessorIntegrationTest.Process5PercentPacketLoss. 
BUG=262
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/379014

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

BUG=
TEST=voe_auto_test on Linux

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

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

BUG=none
TEST=neteq_unittests

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

BUG=114
TEST=audioproc_unittest on all platforms.

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

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

BUG=185
TEST=audioproc_unittest

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

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

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

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

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

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

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

BUG=
TEST=

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

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

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

BUG=
TEST=

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

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

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

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

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

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

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

Fixed errors from GCC 4.6 compilation.

BUG=
TEST=

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Changing PT for CNG-WB to 98

Remove warnings when building NetEQ with NETEQ_DELAY_LOGGING

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Additional change: removed unused function and member from ACMResampler.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Adds a test to the VCM robustness tests.

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

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

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

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

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

TEST=audio_coding_unittests

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

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

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

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

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

BUG=
TEST=Compiling on Linux.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Also adding a new unittest to cover this case.

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

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

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

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

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

BUG=210
TEST=

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Introducing a proper translation between structs NetworkStatistics
and ACMNetworkStatistics.

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-01-04 09:55:09 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
kjellander@webrtc.org
5490c71a1b Converted to gtest, writing output files properly and no longer uses exceptions. 
This test now runs and fails as a gtest should (previously it always
exited with 0 even if the tests failed).
The audio_coding_module_test target no longer uses exceptions in the generated project.
Output files are written to our global output folder, using
testsupport/fileutils.h.

BUG=
TEST=audio_coding_module_test on all platforms, in Debug+Release

Review URL: http://webrtc-codereview.appspot.com/334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 13:34:18 +00:00
stefan@webrtc.org
898f881e32 Make sure the next frame to be decoded is cleaned up if it's empty. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 09:13:14 +00:00
niklas.enbom@webrtc.org
6c9be123ef Letting strncpy do its job. Landing and extending http://webrtc-codereview.appspot.com/329010/ on behalf of tbreisacher. 
Review URL: http://webrtc-codereview.appspot.com/335009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:59:31 +00:00
stefan@webrtc.org
8c5d24266e Fix VP8 layer 2 sync dependencies. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/333010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:56:04 +00:00
henrik.lundin@webrtc.org
00e730730e Refactoring RtpFormatVp8Test 
This is the first change in a series of changes to get new functionality
into the VP8 packetizer.

This first refactors the RtpFormatVp8Test class, without changing the
operation of the tested RtpFormatVp8 class. A test helper class
RtpFormatVp8TestHelper is introduced to reduce code duplication.

Review URL: http://webrtc-codereview.appspot.com/304009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-21 08:51:36 +00:00
mikhal@webrtc.org
61045a4a03 video_coding/jitter_buffer: Account for layer info when searching for the next frame 
Review URL: http://webrtc-codereview.appspot.com/328003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 23:24:19 +00:00
andrew@webrtc.org
a38ce09919 Fix last Mac/clang compile error. 
Fixes "receiver is a forward class and corresponding @interface may
not exist" error.

TEST=build on Mac with -Werror enabled.
TBR=zakkhoyt@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/333012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 22:23:21 +00:00
pwestin@webrtc.org
061fa5b828 Changed handling of padding data. 
Review URL: http://webrtc-codereview.appspot.com/331008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:56:17 +00:00
henrik.lundin@webrtc.org
dbba1f969f Packet waiting-time statistics 
Adding new statistics API to NetEQ, reporting the waiting time
for each frame. The output is raw waiting time for the frames
that have been decoded since the last statistics report (or
maximum 100 frames). The statistics are reset on each query.

Implemented functionality in ACM to query NetEQ for the raw
waiting times, and process it to produce max, average and
median.

Updating common_types.h and VoiceEngine tests to include the
new metrics.

Unit tests are also added for NetEQ and AcmNetEq.

Review URL: http://webrtc-codereview.appspot.com/328011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:45:05 +00:00
henrik.lundin@webrtc.org
219acc6cec Including Brighten function in namespace VideoProcessing 
This change is in response to Issue 173.

BUG=http://code.google.com/p/webrtc/issues/detail?id=173

Review URL: http://webrtc-codereview.appspot.com/328012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 15:33:49 +00:00
stefan@webrtc.org
62fdc42e9c Fix build issue with clang. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/330009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:27:09 +00:00
stefan@webrtc.org
8dc9e4760e Fixes for selective NACKing. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/332007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:12:50 +00:00
tina.legrand@webrtc.org
5efcad1758 We used the wrong syntax for "new", which generated a warning/error building with clang. 
Review URL: http://webrtc-codereview.appspot.com/336003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-20 09:05:55 +00:00
mikhal@webrtc.org
0e7d9d862a Adding layer info consideration when applying FEC protection. In this first version, we hard code protection zero for non-base layer frames. As a future enhancement, an array should be passed from mediaOpt to set the protection per layer. A TODO was added in MediaOpt. 
Review URL: http://webrtc-codereview.appspot.com/330005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 19:04:49 +00:00
mikhal@webrtc.org
190e88a6d3 video_coding: When in hybrid mode, don't NACK non-base layer packets 
Review URL: http://webrtc-codereview.appspot.com/334002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:57:14 +00:00
mikhal@webrtc.org
884d8e7f4b video_coding: Updating sync state based on the layer flag 
Review URL: http://webrtc-codereview.appspot.com/333004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 18:53:05 +00:00
henrik.lundin@webrtc.org
303158588b Revert "Inject TickTimeInterface into VCM and tests" 
This CL reverts r1220.

Review URL: http://webrtc-codereview.appspot.com/336002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:55:45 +00:00
henrika@webrtc.org
e32c08a5a6 Removes usage of default parameters and fixes a bug which was found 
using Clang on Linux.

BUG=none
TEST=none
TBR=pwestin

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 17:39:48 +00:00
stefan@webrtc.org
b33f9dccd6 Correction to how the VP8 wrapper generates picture ids. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/329006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 10:24:02 +00:00
tina.legrand@webrtc.org
398af2337b Solving issue 178, errorbuild warnings on Mac. 
This CL continues the work of solving issue 178. A small change in one file.
Review URL: http://webrtc-codereview.appspot.com/330006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-19 07:36:07 +00:00
henrike@webrtc.org
cf5bcd1fd2 Removed usage of the deprecated critical section constructor in audio_conference_mixer. 
Review URL: http://webrtc-codereview.appspot.com/320007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 23:00:30 +00:00
andrew@webrtc.org
8a44259ea8 Move static consts out of class. 
Still causing a gtest error on non-Win platforms. This should fix it...

TBR=asapersson@webrtc.org
TEST=build on Linux

Review URL: http://webrtc-codereview.appspot.com/332006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 21:24:30 +00:00
andrew@webrtc.org
41192469f6 Switch enums to consts to fix gtest error. 
TBR=asapersson@webrtc.org
TEST=build on Windows

Review URL: http://webrtc-codereview.appspot.com/330008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 20:55:46 +00:00
henrike@webrtc.org
105e07193e Removed usage of the deprecated critical section constructor in modules/utility. 
Review URL: http://webrtc-codereview.appspot.com/321006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 19:53:46 +00:00
marpan@webrtc.org
57353a33f1 FEC Receiver: Fix to how old packets (e.g., re-tranmitted packets in hybird NACK-FEC mode) are treated. 
This change avoids having old packets end up on the current packet list for FEC decoding, and so they are immediately sent out to jitter buffer.
The current list of packets for FEC decoding are sent out only when new packet arrives (with time-stamp greater than current).
Review URL: http://webrtc-codereview.appspot.com/322009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 17:21:09 +00:00
henrik.lundin@webrtc.org
e7d8c56c56 Fix for dual decoder in VCM receiver 
In VCMReceiver::FrameForDecoding, one of the if-cases could sometimes
extract an incomplete frame without first copying the state to the
dual decoder.

Review URL: http://webrtc-codereview.appspot.com/328006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 15:40:52 +00:00
henrik.lundin@webrtc.org
a70f945086 Inject TickTimeInterface into VCM and tests 
The purpose of this change is to introduce dependency injection
of the timer into the video coding module.

Review URL: http://webrtc-codereview.appspot.com/332003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 14:40:05 +00:00
asapersson@webrtc.org
5249cc8f77 Review URL: http://webrtc-codereview.appspot.com/295010 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 14:31:37 +00:00
tina.legrand@webrtc.org
9775a30859 Added variable to catch return value. 
Review URL: http://webrtc-codereview.appspot.com/329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 11:15:46 +00:00
kjellander@webrtc.org
08dec7f449 Now using fileutils.h OutputPath to write output to the right directory and ResourcePath to read resource files from the resource bundle. 
Removed some Valgrind warnings by closing output files. There are still some Valgrind warnings left, that needs to be fixed by a developer with more insight.

Updated all include paths to contain full paths to header files.

Tested in Debug+Release on Linux, Mac and Windows.
All tests ran successfully except the VideoProcessingModuleTest.ContentAnalysis test that fails on Windows with the following error:
unknown file: error: SEH exception with code 0xc0000005
thrown in the test body.
Fixing that is out of scope for this CL.

Review URL: http://webrtc-codereview.appspot.com/266011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 10:31:38 +00:00
tina.legrand@webrtc.org
554ae1ad4e Changes to solve warnings on Mac, issue #178. 
Review URL: http://webrtc-codereview.appspot.com/320005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-16 10:09:04 +00:00
henrike@webrtc.org
7136990a3f Removed usage of the deprecated critical section constructor in udp_transport. 
Review URL: http://webrtc-codereview.appspot.com/321005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 19:17:28 +00:00
leozwang@webrtc.org
0c839fe873 Add new source file to makefile 
Review URL: http://webrtc-codereview.appspot.com/322015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 19:10:24 +00:00
henrik.lundin@webrtc.org
0a10e3c4b2 Fix order of include and guard in tick_time_interface.h 
Review URL: http://webrtc-codereview.appspot.com/331002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 16:08:36 +00:00
henrik.lundin@webrtc.org
c74b2861f3 Fix the include in fake_tick_timer_interface.h 
The include was in error.

Review URL: http://webrtc-codereview.appspot.com/330002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 11:28:31 +00:00
kma@webrtc.org
ee36b9587d corrected android makefile for isac build. 
Review URL: http://webrtc-codereview.appspot.com/321013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 00:18:45 +00:00
andrew@webrtc.org
59ccd5c71f Rename _windows.h -> _win.h in system_wrappers. 
- Also rename _dummy -> no_op which states its purpose more clearly.
- Always use exclusion lists (i.e. sources! instead of sources)

TEST=builds and passes system_wrapper_unittest on Linux, Mac, Win

Review URL: http://webrtc-codereview.appspot.com/317007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-15 00:17:43 +00:00
kma@webrtc.org
6a17340db5 Review URL: http://webrtc-codereview.appspot.com/318014 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 22:16:57 +00:00
kma@webrtc.org
a30093bb85 Added one file associated with check in in r1192. 
Review URL: http://webrtc-codereview.appspot.com/320012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 19:57:23 +00:00
leozwang@webrtc.org
9aa9f44ebc Add new source files because of r1174 
Review URL: http://webrtc-codereview.appspot.com/320011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 19:08:33 +00:00
kma@webrtc.org
f0a964dc0a Optimized WebRtcIsacfix_NormLatticeFilterMa() function for iSAC fix for ARM Neon 
architecture with intrinsics and assembly code. The total iSAC codec speech improved
about 3~5%.

Notes
(1) The Neon version after this optimization is not bit-exact with the generic
C version. The out quality, however, is not worse as verified by test vectors ouput,
and undertandably in theory (32bit x 32bit in Neon is more accurate than the approximation
C code in the generic version).
(2) In Android, a isac neon library will be built. Along with some new function structures,
it is partly for preparation of introducing a run time detection of Neon architecture soon.
Review URL: http://webrtc-codereview.appspot.com/268016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:59:43 +00:00
kma@webrtc.org
6601902504 Introduced WebRtcSpl_SatW32ToW16 to iSAC fix, for Android platforms. 
Review URL: http://webrtc-codereview.appspot.com/315005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:41:07 +00:00
leozwang@webrtc.org
f147bbc878 Change codec test app lib dependency from webrtc lib to codec library 
Review URL: http://webrtc-codereview.appspot.com/317009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 18:22:41 +00:00
henrike@webrtc.org
7cdcde3460 Removed usage of the deprecated critical section constructor in media_file. 
Review URL: http://webrtc-codereview.appspot.com/321004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 17:27:58 +00:00
stefan@webrtc.org
780a07a843 Fix infinite loop bug introduced in r1174. 
Merges CleanUpSizeZeroFrames with CleanUpOldFrames, and changes the
behavior to go through all frames looking for empty frames.

TBR=mikhals

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 15:55:19 +00:00
pwestin@webrtc.org
9fe3d51372 Set the new layer sync bit in the VP8 info struct. 
Review URL: http://webrtc-codereview.appspot.com/324010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 15:13:04 +00:00
henrik.lundin@webrtc.org
fbf5af444b Adding a mockable wrapper class for TickTime in VCM 
The class is called TickTimeInterface, with a fake implementation in FakeTickTime.

Review URL: http://webrtc-codereview.appspot.com/323012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 10:36:10 +00:00
stefan@webrtc.org
ef5247b5b1 Fix session_info_unittest error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/324009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 10:25:38 +00:00
stefan@webrtc.org
0c40d3315f Fixes an assert triggered in jitter_buffer_test and disables deblocking. 
When deblocking is enabled the first frames can include uninitialized
memory. Disabling for now.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/320010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 09:39:30 +00:00
andrew@webrtc.org
6d609b59f3 Fix crashes due to static_instance. 
- Initialize a needed critsect in the constructor of
  UdpSocket2ManagerWindows.
- Don't return NULL when creating a static instance.

TEST=voe_auto_test on Windows.

Review URL: http://webrtc-codereview.appspot.com/324008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-14 02:36:33 +00:00
andrew@webrtc.org
5ae19de3ec Fix error in RtpDump::Start due to r1156. 
- r1156 fixed a check on the _text member of FileWrapper. Turns out this
  was incompatibile with the RTP dumps, which want to write both binary
  and text data. Writing text data to a file open as "b" isn't actually
  an error, so I simply removed the check.
- Also cleans up the interface, most notably removing all WebRtc types.

TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/317005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 22:59:33 +00:00
mikhal@webrtc.org
832cacacff video-coding: Adding a decoded state to the JB logic (JB refactor). 
This new class stores the last decoded info, including temporal info. 
Review URL: http://webrtc-codereview.appspot.com/300005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 21:15:05 +00:00
henrike@webrtc.org
65573f2922 Removed usage of the deprecated critical section constructor in rtp_rtcp. 
Review URL: http://webrtc-codereview.appspot.com/315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 19:17:27 +00:00
stefan@webrtc.org
f4c8286222 Pass NACK and FEC overhead rates through the ProtectionCallback to VCM. 
These overhead rates are used by the VCM to compensate the source
coding rate for NACK and FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/323003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 15:38:14 +00:00
henrik.lundin@webrtc.org
1ced840893 Fixing a nit in the unittest 
This caused some of the build bots to fail.

Review URL: http://webrtc-codereview.appspot.com/324005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 14:59:00 +00:00
henrik.lundin@webrtc.org
eda86dc76b Adding a LayerSync bit to VP8 RTP header 
Updated RtpFormatVp8, ModuleRTPUtility, VP8Encoder and VP8Decoder
to support a new LayerSync ("Y") bit. Note, in VP8Encoder the bit
must be used together with a non-negative value for temporalIdx.
Fixing the plumbing between RTP module and and from VP8 wrapper.
Updating unit tests; all pass.

The new bit is yet to be used by the VP8 wrapper.

Review URL: http://webrtc-codereview.appspot.com/323008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-13 14:11:06 +00:00