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Author SHA1 Message Date
phoglund@webrtc.org
693240f2d9 Fixed many formatting and indentation problems in voe_auto_test. 
Fixed indentation. There is nothing but whitespace changes here, except for removing some extraneous semicolons in .h files and fixing a spelling error in a comment.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 12:32:58 +00:00
henrik.lundin@webrtc.org
598ad06432 Fixing compiler warning in NetEQ 
With some compiler settings, a warning was issued for NetEQ,
saying that pw16_randVec was accessed out of bounds.
This did never happen in practice, but this change makes the
compiler understand this.

Review URL: http://webrtc-codereview.appspot.com/309001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 11:52:09 +00:00
stefan@webrtc.org
b3bd1cd5f1 Fixes Valgrind warnings in the default VCM tests. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 11:48:09 +00:00
henrik.lundin@webrtc.org
bf86c33b0e Removing OutputDebugString from rtp_rtcp module 
This is in response to WebRTC issue 167.

BUG=http://code.google.com/p/webrtc/issues/detail?id=167

Review URL: http://webrtc-codereview.appspot.com/301013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:44:05 +00:00
henrik.lundin@webrtc.org
44ef3774ce Fixing a compiler error in NetEQ 
This error would only arise when compiling without support for
DTMF (which is not the default config).

Review URL: http://webrtc-codereview.appspot.com/310001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:43:25 +00:00
phoglund@webrtc.org
5b343aedcc Added missing .h files to .gypi files so they will show up in xcode / vc projects. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 10:34:35 +00:00
stefan@webrtc.org
58927e8d8f Disable deblocking temporarily due to Valgrind warnings. 
Also corrects the copying of the decoded image data for frames
with odd width or height.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 08:13:31 +00:00
marpan@webrtc.org
1d34212a45 FEC: Update to packets masks (FEC generator matrix) in fec_private_tables.h 
A set of the packet masks (up 10x10 size) are modified for the following reasons:

1) have more even column and row degree (number of 1 bits), when possible.

2) if cases where the column degree cannot be constant across source packets, placed the extra 1 bit in the first packet column (so little more protection on 1st partition), as opposed to having some ~middle source packet have the extra bit.

3) in some cases, made the mask a little more sparse/reduced the overlap.

Overall the average recovery is a little better with these masks.

Mask sizes above 10 will be updated in future changelist.
Review URL: http://webrtc-codereview.appspot.com/305001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 18:50:11 +00:00
kma@webrtc.org
4a8b1eaf6e In NS, replaced a divide calculatoin by shifting, and thus saved the MIPS by 5%(ARMv7) and 10%(ARMv7-Neon). Bit is not exact with the original. Quality is similar. 
Review URL: http://webrtc-codereview.appspot.com/298004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 18:04:48 +00:00
henrik.lundin@webrtc.org
b6e58eb5a1 Fix formatting of rtp_format_vp8* 
Sorting out all lint issues and fixing indentation.

Review URL: http://webrtc-codereview.appspot.com/301011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 15:56:18 +00:00
stefan@webrtc.org
c7e2bffb66 Fix header/lib mismatch caused by a constant not defined for header file. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=170
TEST=

Review URL: http://webrtc-codereview.appspot.com/300008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 13:44:25 +00:00
phoglund@webrtc.org
048b037342 Fixed vie_auto_test shutdown race conditions. 
Fixed a race condition crash in vie_auto_test shutdown. Certain tests did not clean up the voice engine properly which caused crashes during certain uncommon timing conditions.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/307001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 10:42:05 +00:00
xians@webrtc.org
eff3c8905f this patch fixes the valgrind warnings in the adm api test for pulseaudio in linux. 
Review URL: http://webrtc-codereview.appspot.com/301012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 10:02:56 +00:00
mikhal@webrtc.org
cae01010bd libyuv unit test: adding check for fread return value 
Review URL: http://webrtc-codereview.appspot.com/303007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-06 00:44:16 +00:00
mikhal@webrtc.org
a5e980a906 Updating jitter buffer test following latest changes. 
Review URL: http://webrtc-codereview.appspot.com/294002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 18:27:31 +00:00
phoglund@webrtc.org
23e1c0a0b1 File handling in vie_auto_test now uses fileutils so input and output file end up in a good place. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 15:27:04 +00:00
perkj@webrtc.org
ec7759a8c4 Fix broken vie_capture_module_test on mac. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/303006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 12:17:10 +00:00
perkj@webrtc.org
8627adc158 Refactored Video capture Unit test to use gtest. 
Fix Valgrind warnings on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 09:58:55 +00:00
stefan@webrtc.org
0ae71b9ccb Disable temporal layers when building with Chromium. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 08:42:52 +00:00
henrika@webrtc.org
af71f0e5d9 Fixes two minor issues reported by the Coverty Integration Manager. 
BUG=none
TEST=voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/302002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-05 07:02:22 +00:00
andrew@webrtc.org
c9cc3750cf Add missing system_wrappers dependency. 
TBR=kma@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/301009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 20:51:20 +00:00
kma@webrtc.org
b59c031660 For Android ARMv7 platforms, added a feature of dynamically detecting the existence of Neon, 
and when it's present, switch to some functions optimized for Neon at run time.
Review URL: http://webrtc-codereview.appspot.com/268002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 18:34:50 +00:00
andrew@webrtc.org
ae7017d588 Fix missing dependency in audioproc. 
TBR=bjornv@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/300006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 01:43:29 +00:00
andrew@webrtc.org
7bf2646e4d Make protobuf use optional. 
- By default, disable the AudioProcessing protobuf usage in the Chromium
  build. The standalone build is unaffected.
- Add a test for the AudioProcessing debug dumps.

TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/303003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-03 00:03:31 +00:00
mflodman@webrtc.org
626fbfd4cd Correcting vie_encoder nits. 
Review URL: http://webrtc-codereview.appspot.com/302004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 23:39:11 +00:00
perkj@webrtc.org
6b1bfd6c5e Changed webrtc::ACMCodecDB::neteq_decoders_ to a const array. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 12:48:19 +00:00
pwestin@webrtc.org
db221d2b81 Fixes to temporal layers, Henrika please review src/common_types.h 
Review URL: http://webrtc-codereview.appspot.com/286001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 11:31:08 +00:00
phoglund@webrtc.org
6aed73d218 Fixed release compilation error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 11:14:12 +00:00
henrik.lundin@webrtc.org
e26aad4a9e Disable NetEQ unittest for Windows 
Disable NetEqDecodingTest::TestNetworkStatistics for Windows.
It was never tested for Windows. Something is causing it to
fail, probably need different set of test vectors.

Review URL: http://webrtc-codereview.appspot.com/302003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:27:14 +00:00
stefan@webrtc.org
9cb2b56b65 Corrected a fread verification. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/301006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:22:29 +00:00
phoglund@webrtc.org
b956b4856a vie_auto_test may now be run in automated mode on all three platforms. 
Fixed chrash bug on Mac, but there are still crash bugs since a couple weeks back. These will have to be fixed separately.

Removed dialogs from capture tests on Windows.

Removed some dead code related to answer files.

Added the last Windows fixes.

Fixed the Mac vie_auto_test runner - it will now run on Mac again. It will still crash randomly on codec and rtcp tests though.

Fixed compilation error.

Got patch to commit on Mac.

Temp commit on mac

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/292011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 10:19:27 +00:00
perkj@webrtc.org
38ca4f2953 Fix code review comments. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:34:10 +00:00
perkj@webrtc.org
d3eac4158c Fixed webrtc::perm variable. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:34:01 +00:00
perkj@webrtc.org
1b72fcd27b Fix symbol RTPFILE_VERSION. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:33:51 +00:00
stefan@webrtc.org
772d70bcd2 Fix release build error. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 09:18:49 +00:00
stefan@webrtc.org
a4a88f90c4 Implemented NACK based reference picture selection. 
This CL implements NACK based reference picture selection for VP8. A separate
class is used for keeping track of the references and managing the VP8 encode
flags. Appropriate tests have also been added.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/284002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 08:34:05 +00:00
henrike@webrtc.org
4b00560a6e Fixes build error in rtp_rtc module introduced in r1076. 
Review URL: http://webrtc-codereview.appspot.com/301005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-02 00:32:24 +00:00
punyabrata@webrtc.org
c1ed87602a Adding some error handling functionality in the windows audio core implementation to 
stop rendering automatically and throw a playout-error callback when RequestPlayoutData
fails
Review URL: http://webrtc-codereview.appspot.com/300003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:55:35 +00:00
mflodman@webrtc.org
c6182915a3 Fix vie_encoder.cc. 
TBR=ajm

Review URL: http://webrtc-codereview.appspot.com/301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:45:28 +00:00
mflodman@webrtc.org
84d17838ac Refactored ViEEncoder. 
Style changes + QT Metrics class from h-file to cc-file, type changes will be in another CL.

Review URL: http://webrtc-codereview.appspot.com/303001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 17:02:23 +00:00
kjellander@webrtc.org
5f4f69ac57 Removing sleeps from vp8_test. 
These sleeps were remains from earlier tests that required them to work with some codecs. Removing these sleep calls cut the execution time from 90s to 30s on my machine.

Review URL: http://webrtc-codereview.appspot.com/304004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:50:04 +00:00
pwestin@webrtc.org
0644b1dc35 Introduce a mockable RtpRtcpClock interface replacing ModuleRTPUtility time functions 
A new RtpRtcpClock interface has been added to rtp_rtcp_defines.h
and provides time facilities used by an RTP/RTCP module. Also,
NTP constants have been made public in the
webrtc::ModuleRTPUtility namespace to make implementation of
external clocks easier.

An overloaded version of CreateRtpRtcp() accepts a clock argument. By
default, if no clock is provided, the module uses the system clock
(old ModuleRTPUtility implementation).

Throughout the RTP/RTCP module code, calls to TickTime and
ModuleRTPUtility time functions have been replaced with calls to time
methods on a clock object.

The following classes take a clock object in their constructor and
hold a _clock field (either directly, or inherited from a parent):

Bitrate
ModuleRtpRtcpImpl
RTCPReceiver
RTCPSender
RTPReceiver
RTPSender
RTPSenderAudio
RTPSenderVideo

Methods from other classes that do not derive any of those and
require a time take an additional nowMS parameter, that should be
the result of calling GetTimeInMS() on a clock object.
Review URL: http://webrtc-codereview.appspot.com/268017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:42:31 +00:00
bjornv@webrtc.org
132feb1270 Made tables static. 
In this CL global tables have been moved to where they are actually used. If for some reason they need to be available in a larger scope we can add them again at that point.
Review URL: http://webrtc-codereview.appspot.com/303002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:40:50 +00:00
kjellander@webrtc.org
4c4b7f500f Converting vp8_test to use fileutils and gtest 
Review URL: http://webrtc-codereview.appspot.com/289012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 15:24:36 +00:00
tina.legrand@webrtc.org
f64162c335 Adding const to a number of constant tables. Setting some tables to static. 
Patch set 2: Renaming static const tables. They no longer need the prefix WebRtc_Isac...
Review URL: http://webrtc-codereview.appspot.com/301001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 13:01:39 +00:00
bjornv@webrtc.org
bedabb25bf Added const on const tables. 
Builds on Linux.

Tommi: Can you try on Windows?
Review URL: http://webrtc-codereview.appspot.com/300002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-01 07:44:32 +00:00
henrike@webrtc.org
c2ac8953d5 Fixes Valgrind warnings in system wrappers unittest. 
Review URL: http://webrtc-codereview.appspot.com/293006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 22:46:59 +00:00
zakkhoyt@webrtc.org
a7e70b43e2 When entering fullscreen mode, the CocoaRenderView is attached as a subview to a new full screen window. 
When the class is torn down, the view was not being attached back to it's original NSView. I added a 
new class variable to remember the original superview and then reattach it at the appropriate time.
Review URL: http://webrtc-codereview.appspot.com/290009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 22:30:50 +00:00
mikhal@webrtc.org
b9db43e1b6 video_coding/jitter buffer: Reduce delay on a complete frame: No need for the next frame when current frame is already complete. 
Review URL: http://webrtc-codereview.appspot.com/289007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:38:01 +00:00
mflodman@webrtc.org
511f82eee9 Refactored ViESyncModule. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/291007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:31:36 +00:00
perkj@webrtc.org
68f2168978 Remove global voe::Channel::numSocketThreads. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:11:23 +00:00
mflodman@webrtc.org
27a82a65ca Refactored ViEBaseImpl. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/290008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 18:04:26 +00:00
andrew@webrtc.org
587c844741 Query the capture volume immediately on Win Core. 
This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.

Review URL: http://webrtc-codereview.appspot.com/297003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 17:43:05 +00:00
henrik.lundin@webrtc.org
524eb48081 Removing deprecated NetEQ APIs 
Removing WebRtcNetEQ_GetPreferredBufferSize and
WebRtcNetEQ_GetCurrentDelay and all dependent APIs.

Review URL: http://webrtc-codereview.appspot.com/289006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 16:21:22 +00:00
xians@webrtc.org
0dffc6449a To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory. 
This patch disbale some codecs which are not considered necessary. 
Review URL: http://webrtc-codereview.appspot.com/299001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 15:35:44 +00:00
stefan@webrtc.org
0c2adf0b75 Fix bug introduced when enabling VP8 frame dropping. 
Also fixes two unit test mismatches.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/299002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 14:41:58 +00:00
stefan@webrtc.org
ac2c677bf6 Make all video_coding tests use the resources and output directories. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/298001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 14:23:39 +00:00
andrew@webrtc.org
d2daa5c13e Use clang by default on Mac. 
But disable Chrome clang plugins for the time being.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/297005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 01:16:06 +00:00
andrew@webrtc.org
268257475b Fix one more Objective-C clang error. 
(Analogous to r1056).

BUG=issue78

Review URL: http://webrtc-codereview.appspot.com/297004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-30 00:54:04 +00:00
zakkhoyt@webrtc.org
2687b261d5 Since the CocoaRenderView is forward declared with @class instead of imported, 
instance must be cast to NSView* when passed to NSView's addSubView method.
Review URL: http://webrtc-codereview.appspot.com/288001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 23:55:19 +00:00
punyabrata@webrtc.org
c9801465b6 Adding a check to ensure that the memcpy does not exceed bounds of the arrays. 
Review URL: http://webrtc-codereview.appspot.com/290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:49:54 +00:00
andrew@webrtc.org
1e91693fe2 Move stream_delay check to ProcessStream(). 
- was_stream_delay_set_ was being incorrectly reset in
AnalyzeReverseStream().
- Added tests to catch this case.

BUG=
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/291011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:28:57 +00:00
henrike@webrtc.org
0bf2ca2eed Fixes broken unit test http://code.google.com/p/webrtc/issues/detail?id=154 
Review URL: http://webrtc-codereview.appspot.com/292007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 18:21:46 +00:00
mikhal@webrtc.org
5fef05b529 libyuv: Updating paths for test files 
Review URL: http://webrtc-codereview.appspot.com/289010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 17:50:07 +00:00
mflodman@webrtc.org
ffabb59f6e Refactored ViERefCount. 
In a coming CL: Use ref count in system_wrappers instead of this class.

Review URL: http://webrtc-codereview.appspot.com/291010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 17:31:21 +00:00
henrik.lundin@webrtc.org
fc9b903fbe Enable NetEQ statistics unit testing 
Adding test target NetEqDecodingTest::TestNetworkStatistics.
Update neteq_unittest to get files from resources folder.
Update DEPS file to get resources revision 2.

Review URL: http://webrtc-codereview.appspot.com/291013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 15:38:27 +00:00
henrik.lundin@webrtc.org
2d8125dd1a Testing NetEQ network statistics 
Implementing helper function for new unit test
NetEqDecodingTest::TestNetworkStatistics. The test itself
remains to be defined. (Will be added in a coming CL.)
This change required some refactoring of the test code
to avoid excessive code duplication.

Review URL: http://webrtc-codereview.appspot.com/295009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 14:30:28 +00:00
kjellander@webrtc.org
c625c1010a Updated system_wrappers_unittests to use the test_support_main target. 
Review URL: http://webrtc-codereview.appspot.com/291012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 12:11:06 +00:00
stefan@webrtc.org
932ab18d32 Default to always NACKing residual losses when having both FEC and NACK. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/296002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 11:33:31 +00:00
bjornv@webrtc.org
4b80eb4fcd Name change resampler.c/h to aec_resampler.c/h. 
To avoid possible conflict with resampler in common_audio.
Review URL: http://webrtc-codereview.appspot.com/296006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 08:44:01 +00:00
mflodman@webrtc.org
611e4c3253 Refactored ViEPerformanceMonitor. 
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/289009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 02:39:28 +00:00
mikhal@webrtc.org
a85590d383 libyuv: Adding Android.mk 
Review URL: http://webrtc-codereview.appspot.com/291009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-29 01:42:57 +00:00
mflodman@webrtc.org
ad4ee3659e Refactored ViEReceiver. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 22:39:24 +00:00
marpan@webrtc.org
9d8bec6f76 FEC: Fix to valgrind warning. 
Review URL: http://webrtc-codereview.appspot.com/292009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 22:10:05 +00:00
andrew@webrtc.org
400ad6928e Fix compile warning in NS. 
BUG=issue151
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/290005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 21:33:42 +00:00
marpan@webrtc.org
d1b7932adf VP8: Setting non-zero (conservative) threshold for frame dropper. 
Review URL: http://webrtc-codereview.appspot.com/291001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 19:20:31 +00:00
mikhal@webrtc.org
2cdb2d3833 Adding Libyuv to Webrtc: 
- Adding library to DEPS file
 - Adding Wrapper implementation and tests. 

This is an interim state, as these files are not being linked at this stage.
Review URL: http://webrtc-codereview.appspot.com/259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 18:09:41 +00:00
xians@webrtc.org
e07247af8d Valgrind reports a racing condition on _sending because it is accessed by 
both TransmitMixer::PrepareDemux() and StartSend()/StopSend().
Put a lock to resolve it.
Review URL: http://webrtc-codereview.appspot.com/293005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-28 16:31:28 +00:00
andrew@webrtc.org
1e39bc80dc Handle debug files from multiple AEC instances. 
Review URL: http://webrtc-codereview.appspot.com/295006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-27 23:46:23 +00:00
andrew@webrtc.org
a919d3a643 Don't return a zero delay with insufficient data. 
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.

BUG=
TEST=audiproc, audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/292004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-27 23:40:58 +00:00
stefan@webrtc.org
94a8c03141 Slightly increased bandwidth adaptation at both receive- and send-side. 
The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/297002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 14:09:37 +00:00
xians@webrtc.org
8738d277a1 Valgrind detects that there are racing conditions in RTPReceiver::PacketTimeout and RTPSender 
This CL fixes two of them.
Review URL: http://webrtc-codereview.appspot.com/295005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 13:43:53 +00:00
henrik.lundin@webrtc.org
0fcc2eb368 Cleaning up neteq_unittest 
- Conforming to testing standards.
- Fixing a way of generating new reference output files.
- ifdef the test to run only on linux 64-bit
- Renaming unittest source file.
- Renaming test vectors

Review URL: http://webrtc-codereview.appspot.com/296007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 13:43:42 +00:00
henrik.lundin@webrtc.org
789da89d37 Fix a valgrind warning in NetEQ 
The special cases for packet sizes <= 10 ms (one case for each
sample rate) resulted in reading outside of the pw16_decoded
vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast
gets correct input and output vector lengths.

Review URL: http://webrtc-codereview.appspot.com/295008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 12:35:31 +00:00
stefan@webrtc.org
0ee8ba1929 Remove WebRTC dependency on libvpx_lib and libvpx_include. 
Removes dependencies on libvpx_lib and libvpx_include targets when
building with Chromium.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 12:12:43 +00:00
xians@webrtc.org
83661f534e fixing the racing conditions 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:58:15 +00:00
henrik.lundin@webrtc.org
859626570a VP8 RTP work 
Fixing the plumbing to get the KEYIDX between VP8 wrapper and
rtp_rtcp module. Also fixing a missing pipe for temporalIdx

Review URL: http://webrtc-codereview.appspot.com/295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 10:17:00 +00:00
braveyao@webrtc.org
0a18522e1b Add support to 96kHz sampling rate to Windows CoreAudio interface. 
Review URL: http://webrtc-codereview.appspot.com/295003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-25 02:45:39 +00:00
mflodman@webrtc.org
26b9777e62 Only trigger one call to OnNetworkChanged for each incoming RTCP packet. 
Review URL: http://webrtc-codereview.appspot.com/289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 15:22:33 +00:00
mflodman@webrtc.org
471e83e592 Refactored ViESharedData. 
Only vie_shared_data.* are refactored, all *_impl.cc are only changed due to changed names of members in ViESharedData. These files will be refactored later, so the indentation in these files might be corrupt at this stage.

References are not changed to pointers at this stage.

Review URL: http://webrtc-codereview.appspot.com/292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 15:16:00 +00:00
henrik.lundin@webrtc.org
9af365d3c5 Fixing VP8 RTP parser bug 
Missing one initialization of new struct variable hasKeyIdx.

TBR=stefan@webrtc.org

Review URL: http://webrtc-codereview.appspot.com/296004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 13:28:29 +00:00
henrik.lundin@webrtc.org
6f2c0168f0 Updating to VP8 RTP spec rev -02 
Updating the VP8 packetizer class (RtpFormatVp8) and VP8 parser
(in class RTPPayloadParser) to follow the -02 revision of the spec.
See http://tools.ietf.org/html/draft-ietf-payload-vp8-02.

Updating the unit tests, too. Finally, updating the tests to
follow the recommendations from the test team; specifically
including the test code in the webrtc namespace, and omitting
the main function at the end of each test file.

Review URL: http://webrtc-codereview.appspot.com/296003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 12:52:40 +00:00
mflodman@webrtc.org
6d26ef76ea Refactored ViESender. 
In a later CL:
- References -> const or ptr.

Review URL: http://webrtc-codereview.appspot.com/291003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 08:31:06 +00:00
kjellander@webrtc.org
d492f72e43 Added empty unit tests to get code coverage measured. 
In order to get code coverage recorded, there must be an executing test that is linked to the code to measure.
These projects are currently not showing up in the code coverage.

Review URL: http://webrtc-codereview.appspot.com/293002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 07:20:00 +00:00
amyfong@webrtc.org
55d81ea517 ViE Custom Call observer now using pointers, fixed protection method and miscellaneous TODO cleanup 
Review URL: http://webrtc-codereview.appspot.com/282004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-24 01:15:10 +00:00
andrew@webrtc.org
ba028a31c9 Fix sample rate printout in process_test. 
TBR=bjornv

Review URL: http://webrtc-codereview.appspot.com/292005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 20:37:12 +00:00
phoglund@webrtc.org
f3d10d3dfd Fixed release compilation error-warnings. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/290004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 15:56:27 +00:00
phoglund@webrtc.org
c4c56ed20b Rewrote vie_auto_test to use googletest macros. 
Removed error counting entirely - that's completely managed by googletest now, except for custom call, loopback and simulcast call.

Rewrote remaining tests to use GTest asserts.

Rewrote more tests to use GTest macros. The External Codec module is now in the build by default.

Merge branch 'master' into macro_improvements

Rewrote some more code to use GTest asserts.

The manual standard tests now also go through gtest.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/287002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 15:23:11 +00:00
bjornv@webrtc.org
48b68c0c24 Added support for 96 kHz sampling frequency. 
Updated resampler_unittests with the new valid combinations.
Verified audio quality on files.

TEST=resampler_unittests, voe_auto_test
BUILDTYPE=Debug, Release
PLATFORM=Linux
Review URL: http://webrtc-codereview.appspot.com/294001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 13:50:41 +00:00
henrik.lundin@webrtc.org
4257790d2d NetEQ-related bug in ACM 
Fixing a bug when creating new NetEQ slave instances in ACM.
The old code called WebRtcNetEQ_GetCurrentDelay() for the
master instance to get a delay value for WebRtcNetEQ_SetExtraDelay().
This is wrong, since WebRtcNetEQ_GetCurrentDelay() reports on the
current total buffer length, while WebRtcNetEQ_SetExtraDelay() is
the extra delay that is desired to in order to sync with video.

The fix includes keeping the extra delay value in a member variable
in the ACMNetEQ class.

Review URL: http://webrtc-codereview.appspot.com/295001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 13:04:05 +00:00