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965 Commits

Author SHA1 Message Date
stefan@webrtc.org
7889a9b49a Remove use of CriticalSectionScoped(CriticalSectionWrapper& critsect) in VCM. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 08:18:24 +00:00
mikhal@webrtc.org
ea71440aec video_coding: Adding the non reference flag to the receive side logic. 
Review URL: http://webrtc-codereview.appspot.com/323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 02:29:34 +00:00
andrew@webrtc.org
114c790be7 Remove character limit in WriteText(). 
- vfprintf can be used directly here, removing the need for the interim
  buffer. This change allows us to remove the artificial character limit.
- Fix bugs with _text. It wasn't actually getting set earlier, and the
  check was wrong.
- Remove asserts that should use real error checks.

TEST=DataLog and VoECallReport (through voe_auto_test), the only users of WriteText().

Review URL: http://webrtc-codereview.appspot.com/323001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 02:33:33 +00:00
henrike@webrtc.org
2f47b5a70f Fixes a build error when disabling trace (which is done when building with chrome flag is set). 
Review URL: http://webrtc-codereview.appspot.com/318006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 00:44:47 +00:00
wjia@webrtc.org
c6b286fc04 add correct include paths for both chrome build and standalone build. 
BUG=none
TEST=compiles
Review URL: http://webrtc-codereview.appspot.com/320008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-10 00:20:54 +00:00
andrew@webrtc.org
bbea716117 Workaround for libyuv libjingle breakage. 
libjingle depends on ConvertFromI420. This was previously available
through vplib. libjingle still has access to the vplib header, but the
implementation is no longer built.

Fortunately, the libyuv wrapper can supply the implementation, if we
hack the signature to return to the unsigned int types. We'll remove
this once libjingle has been updated to use libyuv directly.

Also, roll libyuv to r100 which fixes a gyp warning on Windows.

TEST=build

Review URL: http://webrtc-codereview.appspot.com/323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 19:43:12 +00:00
henrike@webrtc.org
315282c01a Fixes a compiler warning related to dynamically allocated static memory. the fix is to leak the memory since the OS will clean it up anyways. This will not add noise to memory tools so it's ok. The issue is reported here: http://code.google.com/p/webrtc/issues/detail?id=147. 
Review URL: http://webrtc-codereview.appspot.com/267023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 17:46:20 +00:00
mflodman@webrtc.org
d5651b98c5 Refactored ViEFrameProviderBase. 
Only style changes, ointers/references and functions will come in a later CL.

vie_capturer.cc and vie_file_player.cc are only changed du to inheriting protected members from ViEFrameProviderBase.

Review URL: http://webrtc-codereview.appspot.com/324001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 15:20:58 +00:00
xians@webrtc.org
0744ee563d Disable API tests on ALSA since the tests don't work for all the alsa devices. 
Review URL: http://webrtc-codereview.appspot.com/317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 14:05:29 +00:00
henrik.lundin@webrtc.org
6198624815 Remove warnings on Mac (Issue 178) 
Remove an if-else that can never execute the else statement.
Remove double parenthesis.

BUG=http://code.google.com/p/webrtc/issues/detail?id=178
TEST=

Review URL: http://webrtc-codereview.appspot.com/318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 13:58:17 +00:00
mflodman@webrtc.org
5cc4dc9e0c Remove warnings in VideoEngine, capture module and render module. 
BUG=164, 176, 180

Review URL: http://webrtc-codereview.appspot.com/303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 10:12:57 +00:00
mflodman@webrtc.org
b19582b7dc Add pointer constructor to CriticalSectionScoped. 
Mainly added to simplyfy the code, e.g. when having critsect as scoped_ptr in classes.

Review URL: http://webrtc-codereview.appspot.com/302005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 10:02:16 +00:00
henrikg@webrtc.org
af225d6bf6 The change http://webrtc-codereview.appspot.com/299001 (commit 1062) does not do what it intends (exclude codecs from Chromium build). This is a fix for that. webrtc.gyp is not pulled in Chromium, hence it has no effect putting a define there. Moving it to src/build/common.gypi. 
Review URL: http://webrtc-codereview.appspot.com/315002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 09:58:39 +00:00
mflodman@webrtc.org
5885a4162a Refactored ViERenderer. 
Only style changes, function and type changes will come in a later CL.

Review URL: http://webrtc-codereview.appspot.com/321001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 05:52:32 +00:00
mikhal@webrtc.org
2ab104e6be Switching WebRtc to LibYuv. 
General Notes:
1. In general, API structure was not modified and is based on VPLIB. 
2. Modification to API: Return values are based on libyuv, i.e. 0 if ok, a negative value in case of an error (instead of length). 
3. All scaling (inteprolation) is now done via the scale interface. Crop/Pad is not being used.
4. VPLIB was completely removed. All tests are now part of the libyuv unit test (significantly more comprehensive and based on gtest).   
5. JPEG is yet to be implemented in LibYuv and therefore existing implementation remains.
Review URL: http://webrtc-codereview.appspot.com/258001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-09 02:46:22 +00:00
mikhal@webrtc.org
ffa0a9e9c9 updating libyuv to latest version (98). 
This CL also includes some additional adaptations to the code due to the upgrade. 
Review URL: http://webrtc-codereview.appspot.com/306001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 22:45:53 +00:00
mallinath@webrtc.org
7766e2a82d - This issue raised by the chromium team when clang compiler is used. This was not an error as in this case we were accessing IPV6 address with IPV4 struct which is defined as 14 bytes in the header file, but we had the runtime check to determine the address space. 
Now the solution is to use IPV6 structures instead of IPV4 when address space is determined.

I haven't put the new solution behind AF_INET6 flag, as i don't think it's necessary. 
Review URL: http://webrtc-codereview.appspot.com/291014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 21:37:19 +00:00
andrew@webrtc.org
b0be7aa7ae Remove deprecated OS X Core Audio APIs. 
We no longer support the 10.4 SDK, so we can remove the weak-leaking
feature and exclusively use the added-in-10.5 APIs.

BUG=issue143
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/322001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 20:15:36 +00:00
marpan@webrtc.org
63b50f60d6 test_fec: Fix to valgrind warnings. 
Review URL: http://webrtc-codereview.appspot.com/304002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 19:05:39 +00:00
mikhal@webrtc.org
f5ee1dc3e6 video_coding: Adding temporal layer info support to receive side 
Review URL: http://webrtc-codereview.appspot.com/303005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 19:04:47 +00:00
xians@webrtc.org
832d7c6000 Disable typing detection for chromium since CGEventSourceKeyState is violating chromium sandbox. 
Review URL: http://webrtc-codereview.appspot.com/320003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 16:45:46 +00:00
phoglund@webrtc.org
dd094fd6ae Started extracting methods out of the main test. 
Started extracting methods out of the main test, which will hopefully make us able to make the tests independent.

Merge branch 'master' into voe_split_methods

Conflicts:
	src/voice_engine/main/test/auto_test/voe_extended_test.cc
	src/voice_engine/main/test/auto_test/voe_extended_test.h
	src/voice_engine/main/test/auto_test/voe_standard_test.cc
	src/voice_engine/main/test/auto_test/voe_standard_test.h

Extracted methods out of the standard test.

Added space before inheritance colons.

Rolled back some header file changes.

Fixed long lines.

Fixed long lines.

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/313001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 15:07:59 +00:00
henrik.lundin@webrtc.org
d03718d1e4 Use ResourcePath in NetEQ unittest 
Review URL: http://webrtc-codereview.appspot.com/320001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 11:42:43 +00:00
mflodman@webrtc.org
d2ee5d989d Changed sync bug introduced in refactoring. 
Review URL: http://webrtc-codereview.appspot.com/319001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 09:25:11 +00:00
mflodman@webrtc.org
c78209c58b Add log when transport fails to send packet. 
Review URL: http://webrtc-codereview.appspot.com/311002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 09:22:45 +00:00
kjellander@webrtc.org
7de6e10410 Fixing compilation error on Linux 64-bit 
Problem was introduced in http://webrtc-codereview.appspot.com/311001/ because I had projects generated with Valgrind configuration, which is more forgiving about these implicit conversions.

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

Review URL: http://webrtc-codereview.appspot.com/318002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 08:39:13 +00:00
kjellander@webrtc.org
5b97b1216f Splitted FileHandler into FrameReader and FrameWriter classes and moved them to testsupport in test.gyp. 
Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.

Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.

Reworked all includes and GYP include paths to use full directory paths.

Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc

BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/311001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 07:42:18 +00:00
henrike@webrtc.org
441b3fe2a1 Made some global statics have function scope so that the global static count is 0 for the rtp_rtcp module. 
Review URL: http://webrtc-codereview.appspot.com/316001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-08 02:03:49 +00:00
stefan@webrtc.org
cc7b649474 Add trace for the situation when the min bitrate > available bandwidth. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/312001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-07 13:22:06 +00:00
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
kjellander@webrtc.org
543c3eaa46 Fixing Release compilation errors 
Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 12:20:35 +00:00
henrik.lundin@webrtc.org
89ab652250 Cleaning up NetEQ statistics 
Removed struct MCUStats_t and all references to it.
Removed totalDiscardedPackets and totalFlushedPackets
from the PacketBuf_t struct.

Review URL: http://webrtc-codereview.appspot.com/293001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 11:06:05 +00:00
henrik.lundin@webrtc.org
df10de4b27 Removing statistics API from NetEQ 
Removing WebRtcNetEQ_GetJitterStatistics(),
WebRtcNetEQ_ResetJitterStatistics(), and the associated
struct WebRtcNetEQ_JitterStatistics. The change ripples
through all the way to the VoiceEngine API.

Review URL: http://webrtc-codereview.appspot.com/285002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 09:36:23 +00:00
braveyao@webrtc.org
7d3e9498bc This CL is to support certain audio devices which don't offer volume control. Try to be more compatible to those rare cases. 
Review URL: http://webrtc-codereview.appspot.com/276011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 03:35:42 +00:00
mikhal@webrtc.org
2b838b4121 video_coding: updating the session info unit test following recent changes 
Review URL: http://webrtc-codereview.appspot.com/290002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-23 00:20:19 +00:00
mikhal@webrtc.org
425b377973 video_coding: Updating internal_defines to resolve latest build error. Refers to JB flush update. 
Review URL: http://webrtc-codereview.appspot.com/289001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 23:41:29 +00:00
mikhal@webrtc.org
f13388f134 video_coding: Requesting a key frame after a JB flush 
Review URL: http://webrtc-codereview.appspot.com/280006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:57:51 +00:00
mikhal@webrtc.org
6b9a7f8704 video_coding: Allowing for a decodable state independent of selective nacking 
Review URL: http://webrtc-codereview.appspot.com/263001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:48:20 +00:00
andrew@webrtc.org
828af1b4b9 Add lookahead to the delay estimator. 
TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/279014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 22:40:27 +00:00
andrew@webrtc.org
5a529395aa Make DMO init safe when not supported. 
BUG=issue133
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/284001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 18:04:26 +00:00
mflodman@webrtc.org
dfe89e337e Move ViE main/test/AutoTest to test/auto_test. 
Only paths in gyp and mk files are changed, source files are only moved.

Review URL: http://webrtc-codereview.appspot.com/267027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 14:03:15 +00:00
andrew@webrtc.org
8594f7688b Add a gyp variable for AEC debug dumps. 
TEST=process_test.cc

Review URL: http://webrtc-codereview.appspot.com/276012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-22 00:51:41 +00:00
kma@webrtc.org
a249f35203 Correct several makefile errors for Android build. 
Review URL: http://webrtc-codereview.appspot.com/267024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 22:16:10 +00:00
mflodman@webrtc.org
6830bdd929 Fix xcode build. 
Review URL: http://webrtc-codereview.appspot.com/280007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 15:53:15 +00:00
mflodman@webrtc.org
94ea32ef60 Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. 
Review URL: http://webrtc-codereview.appspot.com/283002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 14:49:31 +00:00
kjellander@webrtc.org
274c2efbc1 Adding empty test method required to get code coverage 
Review URL: http://webrtc-codereview.appspot.com/279008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-21 09:36:28 +00:00
marpan@webrtc.org
3caa327af0 VP8 wrapper: Turn on some mild amount of deblocking in post-processing. 
Review URL: http://webrtc-codereview.appspot.com/268015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-19 01:08:09 +00:00
henrike@webrtc.org
ce9d89d892 Fixes linux build error introduced in r980. 
Review URL: http://webrtc-codereview.appspot.com/279012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-19 00:14:37 +00:00
henrike@webrtc.org
ad98a3eed0 Fixes TEST crash triggered by webrtc-codereview.appspot.com/268014. 
Review URL: http://webrtc-codereview.appspot.com/280005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 23:55:12 +00:00
henrike@webrtc.org
31d30700d6 Addressed review comments from http://webrtc-codereview.appspot.com/256004/ 
Review URL: http://webrtc-codereview.appspot.com/256007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 19:59:32 +00:00
kma@webrtc.org
ced118636d Changed keyword __restrict__ to __restrict. 
Review URL: http://webrtc-codereview.appspot.com/279011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 17:51:19 +00:00
henrike@webrtc.org
3798ecb25b Made CPU initialization on Windows lazy to prevent long startup time. 
Review URL: http://webrtc-codereview.appspot.com/268014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 16:25:54 +00:00
kjellander@webrtc.org
543611a77a Reverting r972 due to compilation error on Windows Release build. 
TBR=kma
Review URL: http://webrtc-codereview.appspot.com/282003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 13:25:13 +00:00
bjornv@webrtc.org
2f047ccede Removed unnecessary variable to avoid compiler error on Win. 
Review URL: http://webrtc-codereview.appspot.com/267021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 12:03:25 +00:00
henrik.lundin@webrtc.org
ba74924043 Remove use of exceptions in NetEQ test code 
Replaced the exceptions thrown when codec instance creation failed
with simple exit(EXIT_FAILURE). There is no point in continuing
if creating the codec fails.

Review URL: http://webrtc-codereview.appspot.com/282002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 09:55:01 +00:00
bjornv@webrtc.org
6a9835d59c Delay estimator structural changes. 
Improved the way we handle different data types (float vs fixed) and reduced the complexity by nearly 50%.
We now have a generic struct for both float and fixed delay estimators and a core struct for the binary spectrum based delay estimator. All wrapper codes (for both fixed and float) are gathered in delay_estimator_wrappers.*.
Moved out the far end history buffer to AEC(M).
Added a union to handle difference types when create.
Review URL: http://webrtc-codereview.appspot.com/277004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 08:30:34 +00:00
kma@webrtc.org
fa9b016fb5 Optimized WebRtcIsacfix_AutocorrFix() function for iSAC fix. 
(1) For generic platforms, code was changed to remove the shifting within loops.
Basically, it's just change a loop from
  for() {
    sum += (a*b) >> scale;
  }
to:
  for() {
    sum += (a*b);
  }
  sum >> scale;

Type int64_t is used for sum to make sure no information is not lost.
Performance is about the same as before the change. Bits are not exact,
although in theory the change should have preserved more information. The purpose
of this change is to make the generic code and ARM code bit exact, simpify the code,
while keep the speech quality at least not lower. (Some speech tests might be good.)

(2) For ARM platform, used assembly to optimize the performance. iSAC runs faster
with this change. (Reduced run time of an offline file test from 10.16ms to 8.81ms)
Review URL: http://webrtc-codereview.appspot.com/267014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 02:50:55 +00:00
braveyao@webrtc.org
f556b9d1f4 This modification is supposed to fix the webrtc issue 144/145. With this fix, people could set/get mic volume before StartSend(). 
Review URL: http://webrtc-codereview.appspot.com/277007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-18 02:17:28 +00:00
amyfong@webrtc.org
917fa6b923 ViE Custom Call added SetImageScaleStatus toggle option and other changes. 
1. added SetImageScaleStatus for testing purposes
2. added getting the codec information from the incoming/outgoing stream of a videochannel to print call information
3. fixed problem with toggling the one of the observers
4. did more clean up of the code style (mostly spacing)
5. renamed the GetVideo* functions properly to SetVideo* to reflect what the function does

Currently only tested on mac.  Need to test on win7 & linux before final commit.
Review URL: http://webrtc-codereview.appspot.com/267017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 21:29:39 +00:00
kjellander@webrtc.org
cd7b57ef9e Fixing release compilation error 
Review URL: http://webrtc-codereview.appspot.com/279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 14:26:21 +00:00
kjellander@webrtc.org
3f1cb8e546 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/269018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:56:54 +00:00
kjellander@webrtc.org
cc2ecb3c2e Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/267019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:48:36 +00:00
kjellander@webrtc.org
b72268e147 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/280004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:39:15 +00:00
kjellander@webrtc.org
64a897a772 Restructuring and adding dummy unit test target. 
Empty test added to get code coverage recorded.

Review URL: http://webrtc-codereview.appspot.com/282001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 13:33:11 +00:00
phoglund@webrtc.org
8f89f09626 Note: this patch may seem intimidating but it mostly moves code around and renames things. There are quite few actual changes. 
Separated new-style tests from old-style tests. Abstracted code for reuse.

Fully separated the new automated tests from the old-style tests. We now have old-style tests running in manual mode, old-style tests running in automated mode and new-style tests that uses input files and make actual video comparisons.

Introduced a small "library" of helper functions in order to move a lot
of stuff out of the original base and codec tests, which have been made
dependent on the new "library" (which is a header file and a source
file). The new-style tests also depends on this "library".
The comparison test flags are now required only when the comparison tests actually runs.

Separated comparison tests into its own test since it seems we will be running classic vie_auto_test using a fake video driver on Linux.

Made tbInterfaces follow Google conventions.
Merge branch 'render_to_file' into vivi_driver

Resolution alignment testing is now optional behind a flag.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 10:46:59 +00:00
kjellander@webrtc.org
c05b56a38b Fixing compilation error 
Review URL: http://webrtc-codereview.appspot.com/276010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 08:59:48 +00:00
kjellander@webrtc.org
0403ef419f Restructuring and adding unit test targets on project level instead of in common_audio. 
Review URL: http://webrtc-codereview.appspot.com/280001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-17 08:35:47 +00:00
phoglund@webrtc.org
337dc68992 Included modules in webrtc.gyp and fixed build errors. 
Removed TODO from webrtc.gyp since it is done.

Tabs -> spaces.

Tabs -> spaces.

Tabs -> spaces.

Fixed compilation on Windows.

Added missing file.

Merge branch 'master' into fix_mac_modules

Fixed compilation errors for the modules.gyp on Mac. This included some pretty large refactorings.

 Please enter the commit message for your changes. Lines starting

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 15:36:44 +00:00
niklas.enbom@webrtc.org
af26f64616 Inband DTMF stereo support 
Review URL: http://webrtc-codereview.appspot.com/267011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 12:41:36 +00:00
niklas.enbom@webrtc.org
e33a102eee Resubmitting http://webrtc-codereview.appspot.com/269007/ 
Review URL: http://webrtc-codereview.appspot.com/268012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 10:33:53 +00:00
stefan@webrtc.org
fcf33eb7e0 Limit number of send-side BWE increases to one per second. 
Also report 0 losses if not enough expected packets since
previous receiver report.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/270009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 07:58:31 +00:00
punyabrata@webrtc.org
81d4499dee Microphone volume on Mac not being printed properly due 
to a mismatch in variable type. Additionally, now printing
a volume that will range from 0 - 255
Review URL: http://webrtc-codereview.appspot.com/267016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@951 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-16 02:06:49 +00:00
andrew@webrtc.org
755b04a06e Add RMS computation for the RTP level indicator. 
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 16:57:56 +00:00
andrew@webrtc.org
6a85b17a0a Potential fix for crash after Mac sleep. 
When a Mac goes to sleep, the OS pauses the IO threads. If a
subsequent StopSend/Playout happens, we time out waiting for the IO
threads, but didn't ensure they were shut down.

BUG=
TEST=voe_cmd_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/269013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 16:23:41 +00:00
kjellander@webrtc.org
85596d5bf4 Setting completeFrame to true for all created encoded images. 
Review URL: http://webrtc-codereview.appspot.com/276008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 13:45:25 +00:00
tommi@webrtc.org
cde1e7f42a Use a TraceNoop instance when tracing disabled (to be used in Chromium). 
I'm also adding an empty implementation for static methods in the Trace
interface since the default implementation relies on TraceImpl.
Review URL: http://webrtc-codereview.appspot.com/267013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 12:23:36 +00:00
henrik.lundin@webrtc.org
bc91d5af86 NetEQ tests 
Adding capability to parse RED payloads to the RTPanalyze tool.
Also adding a method to scramble an RTP payload (currently not
used).

Review URL: http://webrtc-codereview.appspot.com/276006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 10:16:01 +00:00
mflodman@webrtc.org
a02ef1ace2 Fix broken tree. 
Review URL: http://webrtc-codereview.appspot.com/267015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 07:50:50 +00:00
mflodman@webrtc.org
1f69c03739 Added size sanity check for copying app specific RTCP data. 
Similar check as done in RTCPUtility::RTCPParserV2::ParseAPPItem.

Review URL: http://webrtc-codereview.appspot.com/277002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-15 06:12:39 +00:00
henrik.lundin@webrtc.org
33df5335bf Change luminance of all pixels by a specified value. 
Modeled on color_enhancement.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/269004
Patch from SriRam <tvnsriram@google.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 15:30:26 +00:00
stefan@webrtc.org
7de07652ad Disables a flaky metric test. 
This is a duplication of issue 255008 since I wasn't able to commit that one
from the computer on which it was created.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/276007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 15:16:16 +00:00
tommi@webrtc.org
ded85f14ef Enable WEBRTC_NO_TRACE for Chromium builds. 
I'm also fixing WEBRTC_TRACE so that it won't break the build but on Linux I had to do something non traditional as is explained in the comments.
Review URL: http://webrtc-codereview.appspot.com/269012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-14 09:39:31 +00:00
andrew@webrtc.org
0db7dc6e18 Add file-playing channels to voe_cmd_test. 
Fix file reading and writing.

TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/279001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-13 01:34:05 +00:00
andrew@webrtc.org
cd8243807e Unpack the full set of audioproc data. 
Review URL: http://webrtc-codereview.appspot.com/276004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 19:13:36 +00:00
kma@webrtc.org
d71d480487 Fixed a build error of audio conference mixer in Android. 
Review URL: http://webrtc-codereview.appspot.com/267009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 17:14:23 +00:00
stefan@webrtc.org
b351d6a8d8 Reverting rev 929 due to failing assert on Linux. 
Failing at: audio_buffer.cc:159

TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/270008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 13:26:05 +00:00
mflodman@webrtc.org
fd3a0efd15 RTP bw estimate fix. 
Review URL: http://webrtc-codereview.appspot.com/279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 10:55:26 +00:00
phoglund@webrtc.org
1144ba2268 Base and codec tests now run verify output and render to file instead of to screen. 
Rewrote the codec test to render to file and do video comparisons.

Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.

Added video analysis to the test. This will make sure that the system output roughly the right thing.

Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.

Made sure no one passes in too large YUV videos into the autotest.

The standard test's output now gets captured for both the left and right windows.

Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/249001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 09:01:03 +00:00
niklas.enbom@webrtc.org
50b3cbe979 First pass. You can now enable a stereo codec and send and receive. This does not include more advances use cases (DTMF etc), but I'd rather keep the CLs manageable. 
Review URL: http://webrtc-codereview.appspot.com/269007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-11 08:31:32 +00:00
kma@webrtc.org
b61c410347 Fixed a couple of Android makefiles to let voe and vie build properly. 
Review URL: http://webrtc-codereview.appspot.com/278001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 18:10:25 +00:00
kma@webrtc.org
13318ef422 (1) Corrected the makefile for testing iLBC in Android, and changed the location of the test makefile to make it consistent with audio_processing. 
(2) Added a makefile for testing fiexed point iSAC in Android.
Review URL: http://webrtc-codereview.appspot.com/266005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 18:00:22 +00:00
mflodman@webrtc.org
7a4eb2837a Calculate the available bandwidth before sending a TMMBR 
Also changed the way TMMBR was processed since it did not match the new bandwidth estimator.

Review URL: http://webrtc-codereview.appspot.com/270003
Patch from pwestin1 <pwestin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:54:46 +00:00
mflodman@webrtc.org
637a59e68e jitter buffer update: waiting for key frame when Nack is enabled and continuity cannot be determined. 
Review URL: http://webrtc-codereview.appspot.com/266010
Patch from mikhals <mikhal@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@924 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 12:50:48 +00:00
tina.legrand@webrtc.org
855a77c972 Audio Coding Module: Fixing a bug that prevented the encoder from being re-initialized when changing codec from mono to stereo. 
Solving issue 130 reported by Niklas.

Reviewer: Turaj
Review URL: http://webrtc-codereview.appspot.com/268007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 08:17:08 +00:00
andrew@webrtc.org
c4f129f97c Improve the mixing saturation protection scheme. 
A single participant is not processed at all. With multiple
participants, we divide-by-2 as before when mixing. Afterwards,
the mixed signal is limited by the AGC to -7 dBFS and then doubled to
restore the original level.

This preserves the level while guaranteeing good saturation protection.

Add a test to voe_auto_test. Hijack and improve the existing mixing test
for this.

TEST=voe_auto_test, voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/241013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@920 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-10 03:41:22 +00:00
andrew@webrtc.org
d30b688751 Remove TraceScan executable. 
Review URL: http://webrtc-codereview.appspot.com/270002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 22:23:20 +00:00
andrew@webrtc.org
4b13fc9c09 Add delay modification to process_test. 
Review URL: http://webrtc-codereview.appspot.com/266007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 19:27:11 +00:00
henrike@webrtc.org
2f32b5c8a7 Fixes an issue where file playing could happen at a lower sampling frequency than the file. 
Details:
The mixer looks at all the participants desired frequency and concludes the highest desired mixing frequency. This is the frequency that the mixer will mix at. Participants that are always mixed are in a separate list and the function concluding the highest desired mixing frequency did not look at that list and therefore always conclude that the lowest mixing frequency is sufficient.
Review URL: http://webrtc-codereview.appspot.com/277003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 19:02:17 +00:00
mikhal@webrtc.org
eb4ef17bbd Removing vplib include and VideoInterpolator when not needed 
Review URL: http://webrtc-codereview.appspot.com/268004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 18:11:02 +00:00
kjellander@webrtc.org
488ed92c3b Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/267003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@912 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:12:40 +00:00
kjellander@webrtc.org
c3a4dcd101 Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/266008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:11:44 +00:00
kjellander@webrtc.org
ad79d6f164 Removing exceptions since not used 
Review URL: http://webrtc-codereview.appspot.com/267002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 16:11:14 +00:00
mflodman@webrtc.org
03a9eb1526 RTP module: Make sure payloadName is null terminated. 
Review URL: http://webrtc-codereview.appspot.com/268006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 14:51:18 +00:00
niklas.enbom@webrtc.org
f3c1b87f00 my eyes started bleeding when I saw this... 
Review URL: http://webrtc-codereview.appspot.com/268005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 12:43:48 +00:00
kjellander@webrtc.org
9dcab8fb14 Restoring Android.mk 
This is the last file left from 256006 that I forgot to restore according to your comments.
The other Android.mk you fixed in 266004.

Review URL: http://webrtc-codereview.appspot.com/268003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 08:59:13 +00:00
niklas.enbom@webrtc.org
4cd841e9a6 Fix win compile error for interpolator_test 
Review URL: http://webrtc-codereview.appspot.com/269003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-09 08:02:16 +00:00
phoglund@webrtc.org
cff98ca6ff Made it possible to run the voe_auto_test standard test in GTest behind a flag. The purpose is to run the whole test without any manual intervention since we want to run the test on a build bot in automated mode. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/267001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 13:08:25 +00:00
henrikg@webrtc.org
c58ef08da2 Removes system CPU measurement for Chrome build. 
It does not work on Chrome Windows, and is anyway not needed for Chrome.
Review URL: http://webrtc-codereview.appspot.com/243006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 08:44:17 +00:00
henrik.lundin@webrtc.org
f15fbc379d Change in RTP module SendVP8 
Changing how the max payload length is calculated. Instead
of handling RTP and FEC header overhead explicitly, call the
MaxDataPayloadLength method which already does it. Avoid redundant code. Had to move MaxDataPayloadLength to the
RTPSenderInterface.

Review URL: http://webrtc-codereview.appspot.com/269002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-08 08:23:47 +00:00
kma@webrtc.org
9b813510eb Changes for building audio coding in anroid. Only makefiles are touched. 
Review URL: http://webrtc-codereview.appspot.com/266004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 23:30:01 +00:00
henrike@webrtc.org
26d3667a26 Fix for broken test after r897 
Review URL: http://webrtc-codereview.appspot.com/274001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 23:24:40 +00:00
henrike@webrtc.org
e2a34f8275 Removes the API for setting RX VAD since the RX vad should always be on anyways. 
Review URL: http://webrtc-codereview.appspot.com/264001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 21:33:24 +00:00
mflodman@webrtc.org
5ae9f5ed6c Adding logs in RTPSender::ReSendToNetwork. 
Review URL: http://webrtc-codereview.appspot.com/273001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 20:03:00 +00:00
kjellander@webrtc.org
bf483844af Restructuring and removing neteq_tests.gypi according to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files. 
I also fixed compilation on Mac (by enabling exceptions for the NetEqTestTools target). Executing the test fails on Mac, but I assume this is because it checks bit exactness, similar to the issue we had with audio_coding_module (see issue 114)

Review URL: http://webrtc-codereview.appspot.com/255004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 16:05:19 +00:00
kjellander@webrtc.org
36e1ad9b5d Restructuring and removing ilbc_test.gypi. 
According to project structure discussed with Andrew. We want to flatten out the hierarchy and minimize the number of GYP files.

No changes at all are being made in the source files; they are just moved.
The only modified files are the GYP file and Android.mk

Kevin: I updated relative paths in Android.mk so please verify it is correct, since I don't know how to build that.

Review URL: http://webrtc-codereview.appspot.com/256006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-07 15:27:11 +00:00
andrew@webrtc.org
b353d21560 ...and now fix the Debug build. 
Review URL: http://webrtc-codereview.appspot.com/272001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-05 00:57:33 +00:00
andrew@webrtc.org
369766ed29 Fix Release mode errors in common_video tests. 
Review URL: http://webrtc-codereview.appspot.com/271001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 23:59:56 +00:00
vikasmarwaha@webrtc.org
a5c4c1f1d4 Fix for WebRTC issue 64, removed the screenupdate thread and events from start render as they are already created in the ctor. 
Review URL: http://webrtc-codereview.appspot.com/253008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 23:22:51 +00:00
marpan@webrtc.org
040cb71e0a Fix windows compilation errors and warning for test_fec. Disabled VERBOSE_OUTPUT. 
Review URL: http://webrtc-codereview.appspot.com/253005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 22:57:56 +00:00
tina.legrand@webrtc.org
731e9aea79 Fixes ACM API test to build on 32-bits machines. 
Changing counters from unsigned int64 to int.
Review URL: http://webrtc-codereview.appspot.com/256010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 07:34:22 +00:00
kjellander@webrtc.org
20a370e875 Changing the namespace of TestSuite to webrtc::test. 
Adding gmock initialization into main test runner class

Review URL: http://webrtc-codereview.appspot.com/254004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-04 01:19:16 +00:00
kjellander@webrtc.org
1a8d08ad76 Changing usage of gtest_main target, to use test_support_main instead. 
Review URL: http://webrtc-codereview.appspot.com/252002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 23:28:47 +00:00
andrew@webrtc.org
89088b963e Fix the path to protoc.gypi. 
It was mistakenly pointing to the trunk/build rather than the
trunk/src/build copy, causing the Chrome build to fail.

TEST=./build/gyp_chromium in Chrome

Review URL: http://webrtc-codereview.appspot.com/253006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 20:43:45 +00:00
tina.legrand@webrtc.org
2475a1953a Committing a file that was part of CL 175002, but for wome reason weren't uploaded correctly. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 17:54:27 +00:00
tina.legrand@webrtc.org
fb389e3b02 This CL is divided in several patches, to make review easier. 
Patch Set 1: Removing blanks at end of lines.
Patch Set 2: Removing tabs.
Patch Set 3: Fixing include-guards.
Patch Set 4-7: Formatting files in the list.
Patch Set 8: Formatting CNG.

Patch Set 9: 
* Fixing comments from code review
* Fixing formating in acm_dtmf_playout.cc
* Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing.
* Refactored constructor of ACMGenericCodec. Rest of file still to be fixed.
* Fixing break; after return ...; in several files.

Patch Set 10:
* Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc
NOTE! Not all files have the right format. That work will continue in separate CLs.

Review URL: http://webrtc-codereview.appspot.com/175002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 17:20:10 +00:00
andrew@webrtc.org
a4b9660372 Add mistakenly removed VAD enabling function. 
This resolves the unknown VAD status warnings introduced in r845.

BUG=
TEST=voe_cmd_test

Review URL: http://webrtc-codereview.appspot.com/252004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 01:36:27 +00:00
mikhal@webrtc.org
e203de7ba2 jitter_buffer updates: 
1. Determining continuity based on pictureId and not seq. numbers when available.
2. Hybrid bug fix: Don't set to decodable when the nack list is empty.
Review URL: http://webrtc-codereview.appspot.com/255001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:42:52 +00:00
pwestin@webrtc.org
7232ad78b2 reverted back the sanity and changed the test 
Review URL: http://webrtc-codereview.appspot.com/254006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:36:32 +00:00
pwestin@webrtc.org
cfc1070586 Fixed sanity for min length 
Review URL: http://webrtc-codereview.appspot.com/259003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-03 00:15:44 +00:00
pwestin@webrtc.org
075e91fa27 Added parsing of width and height from VP8 header 
Review URL: http://webrtc-codereview.appspot.com/241012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 23:14:58 +00:00
henrik.lundin@webrtc.org
679cb07980 Fix build error for release build 
Review URL: http://webrtc-codereview.appspot.com/252003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 19:52:27 +00:00
henrik.lundin@webrtc.org
baf6db5ead Making dual decoder work again in VCM 
Changing the assignment operator in VCMJitterBuffer to a named
function (CopyFrom) instead, since it is not a straight
assignment. Also fixing two bugs in the jitter copy function.

Bug fix in VCMEncodedFrame: The copy constructor did not
copy _length.

In VCM codec database, make sure that the callback object is
preserved when copying back from secondary to primary decoder.

In VP8 wrapper, adding code to copy the _decodedImage to the
Copy() method.

Bugfix in video_coding_test rtp_player:
The retransmissions where made in reverse order. Now new items are
appended to the end of the LostPackets list, which makes the order
correct when retransmitting.

Handling the case when cloning an unused decoder state:
When the decoder has not successfully decoded a frame yet,
it cannot be cloned. A NULL pointer will be returned all
the way out to VideoCodingModuleImpl::Decode(). When this
happens, the VCM will call Reset() for the dual receiver,
in order to reset the state to kPassive.

Review URL: http://webrtc-codereview.appspot.com/239010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 18:58:39 +00:00
kma@webrtc.org
4bb141078f A change to Android makefile for building voe auto test. 
Review URL: http://webrtc-codereview.appspot.com/255007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 16:38:22 +00:00
kjellander@webrtc.org
d292b9c9da Unit tests now compile and run at all platforms. 
Cosmetic changes to mocks.h.

Review URL: http://webrtc-codereview.appspot.com/253003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 16:34:52 +00:00
niklas.enbom@webrtc.org
0ba31331a8 Aligning license file with file header 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 09:31:39 +00:00
henrik.lundin@webrtc.org
895870b68f Adding marker bit to RTPanalyze results 
Review URL: http://webrtc-codereview.appspot.com/254005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-02 08:44:42 +00:00
mikhal@webrtc.org
bb8dfbdee2 updating vpm unit_test following r858 
Review URL: http://webrtc-codereview.appspot.com/255005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 22:07:16 +00:00
turaj@webrtc.org
7395d3d8e9 Addressing issue 115 http://code.google.com/p/webrtc/issues/detail?id=115 
Review URL: http://webrtc-codereview.appspot.com/261002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:33:06 +00:00
turaj@webrtc.org
fac5316856 Address the problem that iSAC could not go 16 kHz. It was addressed in P4 but not moved to svn. 
Review URL: http://webrtc-codereview.appspot.com/261001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:32:46 +00:00
turaj@webrtc.org
9116cf7c9b Have a guard on computing nrg to avoid wrap-around. This is discovered in a release test. During entropy coding of spectrum the value of "nrg" was too large and after shifting it became negative, resulting in decoder error. 
Review URL: http://webrtc-codereview.appspot.com/239016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:29:34 +00:00
mflodman@webrtc.org
29d75b3f7d Only allow increasing capture time. 
Review URL: http://webrtc-codereview.appspot.com/259001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:10:49 +00:00
andrew@webrtc.org
18ee6ec8e9 Use __inline in NS-fixed. 
The use of "inline" was failing to build on Windows.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/255003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:07:46 +00:00
andrew@webrtc.org
3119ecfec8 Fix audioproc build errors on Windows. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/254003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 17:00:18 +00:00
mikhal@webrtc.org
c4ab8706f4 video_processing: Adding logic to avoid a memcpy when not required 
Review URL: http://webrtc-codereview.appspot.com/255002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 16:44:24 +00:00
punyabrata@webrtc.org
0ab521f754 Resolving a crash related to strncopy followed by a strcat 
call. strncopy will not explicity copy or add a "\0" therefore
strcat did not know where to append the "\n" which was causing
an out of bounds crash.
Because we are checking the length, strcpy should be good enough
as it also copies the "\0". Please note that that I am pre-emptively
adding 2 instead of 1 to the length to take into account of the \n
that will be added later.
Review URL: http://webrtc-codereview.appspot.com/253004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 15:19:44 +00:00
kjellander@webrtc.org
d6837709cf Fixing VPMUnitTest compilation error on Windows. 
It tried to include Visual Leak Detector which is not a tool that is installed/configured by default in the build.

Review URL: http://webrtc-codereview.appspot.com/257002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 01:51:10 +00:00
henrike@webrtc.org
b37c628ae4 Fixes crash due to r841. 
Review URL: http://webrtc-codereview.appspot.com/256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 23:53:04 +00:00
kma@webrtc.org
e9f909b575 Move the SetAndroidObjects to VideoCaptureFactory so that ViE can get access to it. 
Review URL: http://webrtc-codereview.appspot.com/244002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 22:24:57 +00:00
andrew@webrtc.org
f1a45d77fb Add missing <stdlib.h> to data_log test. 
BUG=
TEST=system_wrappers_unittests

Review URL: http://webrtc-codereview.appspot.com/256002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:44:54 +00:00
andrew@webrtc.org
3134aacd6b Use fileutils for the audio file in voe_auto_test. 
BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/250010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:31:07 +00:00
kma@webrtc.org
27957508a3 Changed Android makefile to make the lastest video render code run. 
Review URL: http://webrtc-codereview.appspot.com/247005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 21:29:50 +00:00