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1416 Commits

Author SHA1 Message Date
stefan@webrtc.org
1480f02faf Fix VCM test build warnings on Mac with clang. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-12-12 13:45:59 +00:00
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
kjellander@webrtc.org
84736882ad Fixing system_wrappers unittests. 
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 20:44:24 +00:00
andrew@webrtc.org
2c74bab8b9 Remove unneeded assert and tracing. 
This is related to r840.

BUG=
TEST=voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/239019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 19:54:20 +00:00
amyfong@webrtc.org
299e2c9ea4 vie_autotest_custom_call.cc - fixed VieAutotestDevcoderObserver to use const int for videoChannel for IncomingCodecChanged, RequestNewKeyFrame 
- this caused vie_auto_test to fail for Windows (but fine for Linux & Mac).
Review URL: http://webrtc-codereview.appspot.com/253001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 19:10:26 +00:00
kjellander@webrtc.org
177bb523bd Fixing system_wrappers unittests. 
Not complete, but enough to include them in the build again.

Review URL: http://webrtc-codereview.appspot.com/241008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 17:10:01 +00:00
henrike@webrtc.org
066f9e5a2f Ray, please verify that this cl fixes the issue. Once the verification has been made, please review: 
Henrik A: VoE
Andrew: audio_conference_mixer

Thanks!
Review URL: http://webrtc-codereview.appspot.com/241010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 23:15:47 +00:00
henrike@webrtc.org
731ecba47d Review URL: http://webrtc-codereview.appspot.com/251002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 22:49:24 +00:00
braveyao@webrtc.org
1f6d740571 This CL is about to manually reset the ShutdownRenderEvent at StopPlayout(). 
It could happen that if you want to restart playout, the new sponsored Render thread would catch this event
if the previous Render thread quits before this event is set.
With this modification, the device plugging out/in during talking would be supported well.
Review URL: http://webrtc-codereview.appspot.com/248002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 21:30:30 +00:00
wu@webrtc.org
88e0a34815 Remove duplicated code. 
Review URL: http://webrtc-codereview.appspot.com/251001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 17:29:44 +00:00
stefan@webrtc.org
f960211f8b Fixes two jitter buffer bugs related to NACK. 
Avoid decoding delta frames after a Flush() and after requesting
a key frame due to full NACK list.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 16:00:49 +00:00
bjornv@webrtc.org
250cd6f41b Added a VAD unit test to common_audio. At this stage it runs through the API calls, but should later be complemented with tests on a file. 
Review URL: http://webrtc-codereview.appspot.com/243002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:45:58 +00:00
stefan@webrtc.org
eb65860720 Reverts the workaround in r823 and solves a macro bug. 
The macro bug caused frames to be dropped after being grabbed
for decoding.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 12:25:34 +00:00
tina.legrand@webrtc.org
8b1f621e3a Updated gypi for tests to work on osx. 
Review URL: http://webrtc-codereview.appspot.com/250002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 08:57:34 +00:00
amyfong@webrtc.org
ca4666b75c vie wintest added hybrid protection mode 
also fixed Max Framerate to reflect its actually the min framerate
Review URL: http://webrtc-codereview.appspot.com/244010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 21:16:40 +00:00
amyfong@webrtc.org
1e7e60b739 Fixed issue build failling due to vie_autotest_custom_call calling GetBandwidthUsage, which was 
changed in r822.
Review URL: http://webrtc-codereview.appspot.com/240014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 20:53:30 +00:00
amyfong@webrtc.org
51e1bb4e1a vie_autotest_customcall added encoder/decoder observer, maxBitrate set, print call statistics, enable kTraceAll 
When creating a new custom call, now able to set start bit rate (default is 1000)

The following modify call options were added
  9. Toggle Encoder Observer
 10. Toggle Decoder Observer
 12. Print Call Statistics

Also set the trace filter to kTraceAll

File defaults new call VGA (640x480)
Review URL: http://webrtc-codereview.appspot.com/239012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 18:16:35 +00:00
mikhal@webrtc.org
5200a05500 video_coding/jitter_buffer Updating condition on which we return a frame. 
Review URL: http://webrtc-codereview.appspot.com/240011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:54:54 +00:00
mikhal@webrtc.org
30f6376802 VP8: Updating codec version: VP8 version will now return the libvpx version used. 
Review URL: http://webrtc-codereview.appspot.com/247009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:45:00 +00:00
stefan@webrtc.org
2d28aff785 Workaround for an issue where frames are grabbed for decoding prematurely. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:13:18 +00:00
stefan@webrtc.org
fbea4e555d Solves two bandwidth estimation issues and measures the sent video bitrate. 
Issues solved:
1. Possible overflow when reducing the bandwidth estimate at the send-side
2. A burst of loss reports could make us reduce the rate way too far since
   we reduced the rate relative the current estimate and not the actual
   rate sent.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/244011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 16:08:29 +00:00
mflodman@webrtc.org
7e4269e9ee Changed VP8 qp min and added noise reduction. 
Review URL: http://webrtc-codereview.appspot.com/248003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-27 12:59:47 +00:00
mflodman@webrtc.org
8fc663b3ae Don't trigger false ViE SetReceiveCodec warning. 
Review URL: http://webrtc-codereview.appspot.com/250001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 11:30:52 +00:00
kjellander@webrtc.org
6b7799021c Fixing build errors on Windows platform. Minor changes... 
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 02:38:09 +00:00
andrew@webrtc.org
fdde8b3fb7 Add references to src/ copies for LICENSE etc. 
Review URL: http://webrtc-codereview.appspot.com/246007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 01:05:07 +00:00
andrew@webrtc.org
cb18121990 Add an unpacker tool for audioproc debug files. 
- It only unpacks audio data at the moment.
- Also switch to Chrome's protoc.gypi for protobuf targets. This reduces
  the complexity of our targets.

Review URL: http://webrtc-codereview.appspot.com/241009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 00:27:17 +00:00
frkoenig@google.com
fc9bcef8c7 Data alignment fix for SSIM. 
WebRtc_UWord64[2] wasn't always aligned to 128 bytes, which
is necessary for _mm_store_si128.  By declaring the 
variable as __m128i it will always be 128 bytes aligned.

Incorrect include files.

__m128i is defined in emmintrin.h for visual studio.  Extra include on mac and linux is not a problem.
Review URL: http://webrtc-codereview.appspot.com/239013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-26 00:07:32 +00:00
phoglund@webrtc.org
78c767f9ba Rewrote codec test to use fake camera. 
Tests now fail more cleanly if the input video file is incorrect. Fixed some of the style issues in vie_autotest_codec.

Rewrote the automated standard codec test to use the new fake camera.

Started sketching on a new test case. Wrote a new abstraction called ViEFakeCamera which hides the details of how to thread a file capture device in the typical test case.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 12:54:38 +00:00
stefan@webrtc.org
d855c1a4e8 Reverts r807 and fixes the real issue in the VCM. 
This fixes an issue in the VCM where we don't wait for a packet to arrive
if the jitter buffer is empty. This also fixes an issue where an old
packet can trigger a packet event signal for a future frame.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/248001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 11:52:48 +00:00
henrika@webrtc.org
bdb55c806f This CL is an attempt to remove a crash we can see when closing down VoiceEgine. 
It can happen that the capture thread tries to access an invalid object after StopPlayout has been called.

I have also extended the usage of the new ScopedCOMInitializer to all threads. See this step as code cleanup.
Review URL: http://webrtc-codereview.appspot.com/239014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 11:03:28 +00:00
henrika@webrtc.org
a6c23357c0 Solves crash in ADM API unit test for Core Audio on Windows 
Review URL: http://webrtc-codereview.appspot.com/244009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 08:31:33 +00:00
henrika@webrtc.org
5423bc2d0b Adds correct absolute paths to all input files in ADM functional unit tests. 
Files are now read and played out correctly.
Review URL: http://webrtc-codereview.appspot.com/246006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 08:24:20 +00:00
kma@webrtc.org
ca325ececd Corrected a linux build error introduced in issue 246005. 
Review URL: http://webrtc-codereview.appspot.com/246008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 02:36:09 +00:00
wjia@webrtc.org
f0cd394a2e Put fwrite calls under corresponding macros since they shouldn't show up in release build. 
This also make chromeos build happy.
BUG=none
TEST=compile
Review URL: http://webrtc-codereview.appspot.com/247006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:40:43 +00:00
mikhal@webrtc.org
f31826e17b adding a wait on the decode thread when no frames are available 
Review URL: http://webrtc-codereview.appspot.com/246009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:20:54 +00:00
mikhal@webrtc.org
a412924c0e VP8:Setting number of cores based on image size 
Review URL: http://webrtc-codereview.appspot.com/242010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 00:02:43 +00:00
kma@webrtc.org
913644b92d For commiting changes in CL 277002, due to file structure changes introduced during 
the review of the code.
Review URL: http://webrtc-codereview.appspot.com/246005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 21:36:33 +00:00
henrike@webrtc.org
0d0037c2fd Return cached data instead of sleeping in CpuWrapperMac (shaves 2s off WebrtcMediaEngine creation time on Mac). 
Review URL: http://webrtc-codereview.appspot.com/226005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 15:48:14 +00:00
phoglund@webrtc.org
0a9c318c9f The fread result is no longer ignored. 
Changed unsigned longs into uint64_t to be a bit more portable.

Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc

Removed unnecessary use of WebRTC types. Fixed style issues.

Fixed style issues. Added comments where needed.

(After review) Made the standard base test not mirror the render stream since that is assumed to be tested in the render module. Renamed functions accordingly.

Fixed merge errors.
Merge branch 'master' into fake_camera

Conflicts:
	src/video_engine/main/interface/vie_capture.h
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/interface/vie_autotest_defines.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

More updates after review.

Updates after review.

Added new automated test. - Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

Added comments to the new test.

- Fixed a bug which caused test error messages to not get shown.

- Added extended and API tests.
- Abstracted out an integration test base class since all integration
tests set up the exact same way.

- The ViETest::TestError static method will now assert using GTest
asserts if we are running in GTest mode. This gets rid of the hard
asserts that get run otherwise. The hard asserts are still in when using
"classic" mode. TestError will use neither GUnit nor hard asserts if
VIE_ASSERT_ERROR is not defined.
- Formatted vie_autotest_defines.h according to Google style rules.

- Extracted a method for finding a capture device on the system. This
removes a fair bit of logic from the huge test method (mostly straight
statements remain there now).

Rebase from svn.

- Whitespace fixes after review.

Fixed presubmit warning.

- Fixed cpplint.py warnings.

Fixed merge error.

Fixed cpplint.py warnings.

Merge branch 'extended_tests' into fake_camera

Conflicts:
	src/video_engine/main/test/AutoTest/automated/vie_api_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_extended_integration_test.cc
	src/video_engine/main/test/AutoTest/automated/vie_integration_test_base.cc
	src/video_engine/main/test/AutoTest/automated/vie_standard_integration_test.cc
	src/video_engine/main/test/AutoTest/helpers/vie_window_creator.cc
	src/video_engine/main/test/AutoTest/interface/vie_autotest.h
	src/video_engine/main/test/AutoTest/source/vie_autotest_base.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_linux.cc
	src/video_engine/main/test/AutoTest/source/vie_autotest_main.cc
	src/video_engine/main/test/AutoTest/vie_auto_test.gypi

Revert "Revert "- Whitespace fixes after review.""

This reverts commit 3da2a148814e8dea78f73d3feeb32dce690dc2d4.

Revert "- Whitespace fixes after review."

This reverts commit fac670ca313580fb883191ae919091a2637ad0af.

- Whitespace fixes after review.

- Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable. - Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful. - There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts. - Fixed comments for the thread wrapper and the external capture device interface.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

- Fixed cpplint.py warnings.

Fixed presubmit warning.

- Whitespace fixes after review.

Rebase from svn.

- Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

- The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

- Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

- Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 15:33:07 +00:00
andrew@webrtc.org
537096a5c1 Remove unnecessary objective-c compiler flags. 
Review URL: http://webrtc-codereview.appspot.com/239011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 15:07:08 +00:00
phoglund@webrtc.org
c63f788e0f Added fake camera, rewrote one test to use it. 
Wrote a "file capture device" which is a kind of fake capture device. It reads a YUV file from disk and pretends that it is what the "camera" is seeing. This makes is possible to run tests based on video input without having an actual physical camera. This is good because physical cameras are quite unreliable.

Rewrote the standard mirrored preview loopback test so it can use the new file capture device. The old "classic" test is preserved. I tried to minimize duplication between the classic test case and the new one, which turned out to be quite painful.

There are some rough edges left in in the code. Suggested improvements is to get rid of the error counting mechanism since the code seems to assume that TestError invocations cause hard asserts anyway. The code will segfault for certain errors if the hard asserts doesn't happen, which means the error counting mechanism is unnecessary. This, by the way, could be a problem for the new test since it doesn't cause hard asserts.

Fixed comments for the thread wrapper and the external capture device interface.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/224003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 13:20:09 +00:00
henrika@webrtc.org
bf478faebb Ensures that ADM unit tests builds on all platforms. 
Review URL: http://webrtc-codereview.appspot.com/240009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 10:31:02 +00:00
andrew@webrtc.org
f1a605cad6 Update DEPS to support Mac clang build. 
Review URL: http://webrtc-codereview.appspot.com/244003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 15:29:16 +00:00
stefan@webrtc.org
5eb64f06be Fix BitrateSent() API when having a default RTP module. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/242004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:42:50 +00:00
stefan@webrtc.org
158f496030 Fixes a rate control bug in the VP8 wrapper. 
Changes how we signal frame rate and frame durations to the encoder. Rather
than changing the time base, we now only modify the frame durations, while
keeping the timebase constant. The frame duration is currently calculated
from the average input frame rate. Ideally, the frame duration should
be calculated as the timestamp diff, which is the real duration of a
frame, but the encoder doesn't seem to like too varying durations.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/247001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 13:15:16 +00:00
stefan@webrtc.org
ead87b5051 Fix potential issue where frame buffers might be freed while being decoded. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/243004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 06:46:37 +00:00
stefan@webrtc.org
2b0f094c8f Avoid reallocating the decodedImage for every decoded frame. 
Also made sure the right size is allocated.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/240004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 06:39:03 +00:00
mikhal@webrtc.org
ee3dfa6f43 Review URL: http://webrtc-codereview.appspot.com/241007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 00:46:09 +00:00
mikhal@webrtc.org
1af915d8ae video_coding: vp8: Updating error propagation threshold 
Review URL: http://webrtc-codereview.appspot.com/246002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 18:19:18 +00:00
kma@webrtc.org
d75889e2eb Change of Android makefiles to build latest video coding code. 
Review URL: http://webrtc-codereview.appspot.com/239008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 16:28:56 +00:00
henrika@webrtc.org
7cf893743a git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-10-20 12:30:35 +00:00
henrika@webrtc.org
cedbb036d1 [Issue 101] Solves memory leak on Windows 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 12:11:45 +00:00
stefan@webrtc.org
c4d1983b7b Changes in rtp_format_vp8_unittest to match the changes in CL 774. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/241006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 08:19:34 +00:00
mflodman@webrtc.org
ae499a2ac8 Set correct codec info before sending frame to VCM. 
Review URL: http://webrtc-codereview.appspot.com/240003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 05:55:46 +00:00
kjellander@webrtc.org
81f25f9ff8 Fixing build errors on Windows platform. Minor changes... 
Review URL: http://webrtc-codereview.appspot.com/241004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 20:06:56 +00:00
wu@webrtc.org
f3f2f6abdb * Add include_internal_video_capture and include_internal_video_render to include/exclude the internal VCM and VRM. 
* Split the WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER into WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE and WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER.
* Add DummyDeviceInfo for the case when WEBRTC_INCLUDE_INTERNAL_VIDEO_CAPTURE is not defined.
Review URL: http://webrtc-codereview.appspot.com/224005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:42:17 +00:00
henrike@webrtc.org
509c9c5d09 operator + is evaluated before ?: 
Parenthesis ensures the intended behavior.
Review URL: http://webrtc-codereview.appspot.com/239003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:31:01 +00:00
henrike@webrtc.org
4df8c9a2ed Review URL: http://webrtc-codereview.appspot.com/243001 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 18:30:25 +00:00
andrew@webrtc.org
7ecdf585cb Enable chromium_code:1 in the Chrome build. 
Review URL: http://webrtc-codereview.appspot.com/240001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 17:53:56 +00:00
stefan@webrtc.org
ffd28f95c5 Request key frames to battle error propagation. 
The VP8 decoder wrapper will request key frames 30 frames after seeing
a packet loss, if it hasn't received a state refresh (only possible
through key frames in this version).

For this to be possible the jitter buffer has been made aware of
picture ids to be able to detect frame losses. Legacy JB code to
handle streams without marker bits was also removed since it
conflicts with streams with FEC.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/239002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:55:39 +00:00
mikhal@webrtc.org
d0752c370d video_coding: Update to hybrid mode: Set FEC values for zero below a threshold. 
Review URL: http://webrtc-codereview.appspot.com/245001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 15:48:30 +00:00
mflodman@webrtc.org
c693bac6e7 Only start ViEPerformanceMonitor when needed. 
Tested by taking the added part in base extended test and running in Standard test with cpu threashold in ViEPeroformanceMonitor manually changed to 0.

Review URL: http://webrtc-codereview.appspot.com/240005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 13:40:58 +00:00
phoglund@webrtc.org
b5475d0076 vie_auto_test will now obey the Mac .mm rules for files including objective-c code. 
Fixed the Windows build.

Fixed whitespace.

Split the platform-specific code for creating a window manager into separate source files since the mac one must be suffixed .mm and not .cc when we happen to use objective-c code. Tested on Linux.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/214009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 10:59:39 +00:00
bjornv@webrtc.org
4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated. 
Review URL: http://webrtc-codereview.appspot.com/223003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 08:47:40 +00:00
mflodman@webrtc.org
cc412c1735 Remove second instance of ViE PerformanceMonitor. 
Review URL: http://webrtc-codereview.appspot.com/244001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 08:27:30 +00:00
mflodman@webrtc.org
ce8813da4e Using id instead of name when setting Mac/QTKit capture device. 
Review URL: http://webrtc-codereview.appspot.com/241002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 06:45:16 +00:00
andrew@webrtc.org
4d5d5c1267 Reorganize the audio_processing source. 
- Remove main and source directories.
- Change .gyp, .gypi and Android.mk files correspondingly. No other
  source changes.

Review URL: http://webrtc-codereview.appspot.com/241001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 01:40:33 +00:00
andrew@webrtc.org
5d3bdf71ab Fix clang warnings in ViE autotest. 
Review URL: http://webrtc-codereview.appspot.com/239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 01:09:41 +00:00
wu@webrtc.org
8fd93d4d96 Move DeliverCapturedFrame from private to protected. 
Review URL: http://webrtc-codereview.appspot.com/246001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 00:16:36 +00:00
bjornv@webrtc.org
52eddf7378 Made Tina, Andrew and Jan as OWNERS to entire common_audio and removed the sub-OWNERS files. Let me know if that's fine. 
Review URL: http://webrtc-codereview.appspot.com/225006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 07:57:04 +00:00
stefan@webrtc.org
5b15cfc6dd Fix BWE unit test build issue 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 07:22:33 +00:00
kjellander@webrtc.org
61f07c3184 I have made a small fix so it will execute properly from the default working directory location (trunk), finding its resource files. 
The ApmTest.Process test is still failing and needs to be resolved.

BUG=
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Review URL: http://webrtc-codereview.appspot.com/194002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 06:54:58 +00:00
henrik.lundin@webrtc.org
5dedd0ee38 Handling of white-space in DataLog::Combine 
The Combine method cannot handle white-space. Adding a comment to
the header file saying this, and modifying the unittests. Also,
adding a new unittest to test the method.

Review URL: http://webrtc-codereview.appspot.com/217001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 05:45:08 +00:00
amyfong@webrtc.org
929789b528 vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call 
- moved all of the custom call specific functions out of vie_autotest.h and into vie_autotest_custom_call.cc
- added option to modify a running call's video protection method
Review URL: http://webrtc-codereview.appspot.com/234001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@759 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:57:08 +00:00
wu@webrtc.org
76aea651ff When _audioConfigured, should not try to use the _video. 
Review URL: http://webrtc-codereview.appspot.com/224004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:40:32 +00:00
bjornv@webrtc.org
3765bd2cc2 Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests. 
class VoEAudioProcessing
-API renaming:
  SetEchoMetricsStatus() to SetEcMetricsStatus()
  GetEchoMetricsStatus() to GetEcMetricsStatus()
  since delay logging is not strictly an echo metric.
-New API:
  GetEcDelayMetrics()
-Implementations
  --SetEcMetricsStatus() sets same status to all EC related metrics, currently Echo Metrics and Delay Logging.
  --GetEcMetricsStatus() gets an error if all EC related metrics don't have the same status.
  --GetEcDelayMetrics() gets the median and standard deviation of AEC internal delay (on a block by block basis).

class VoECallReport
The changes above leads to changes in the Call Report.
-New API:
  GetEcDelaySummary()
-API updates:
  ResetCallReportStatistics()
  WriteReportToFile()

auto_tests updates:
-Standard test, with new Call Report calls and APM calls
-Extended test, with new Call Report calls and APM calls
Review URL: http://webrtc-codereview.appspot.com/187004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 08:49:23 +00:00
wu@webrtc.org
f10ea31211 Add IncomingFrameI420 to ViEExternalCapture interface to take captured video frame buffer as 3 planes. 
Review URL: http://webrtc-codereview.appspot.com/219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 17:16:04 +00:00
marpan@webrtc.org
14aaaf116a Some re-organization of the fec-uep code: updated protection modes, comments, and some variable/function re-naming. 
Review URL: http://webrtc-codereview.appspot.com/231001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 16:28:02 +00:00
wu@webrtc.org
55c39f0940 Add mallinath@webrtc.org and wu@webrtc.org as the capture owner for US office. 
Review URL: http://webrtc-codereview.appspot.com/230001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 15:34:19 +00:00
wu@webrtc.org
58691ebb97 Remove the DestroyDeviceInfo for mac video capture. (This is missed in r731.) 
Review URL: http://webrtc-codereview.appspot.com/229001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 15:13:16 +00:00
stefan@webrtc.org
d0bdab0128 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. 
Also adding tests for this in vie_auto_test.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/199001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 14:24:54 +00:00
phoglund@webrtc.org
26c041673f Added more tests, fixed a bug and refactored. 
Fixed merge error.

Fixed cpplint.py warnings.

Fixed presubmit warning.

Whitespace fixes after review.

Rebase from svn.

Extracted a method for finding a capture device on the system. This removes a fair bit of logic from the huge test method (mostly straight statements remain there now).

The ViETest::TestError static method will now assert using GTest asserts if we are running in GTest mode. This gets rid of the hard asserts that get run otherwise. The hard asserts are still in when using "classic" mode. TestError will use neither GUnit nor hard asserts if VIE_ASSERT_ERROR is not defined. - Formatted vie_autotest_defines.h according to Google style rules.

Added extended and API tests. - Abstracted out an integration test base class since all integration tests set up the exact same way.

Fixed a bug which caused test error messages to not get shown.

Added comments to the new test.

Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/188002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 13:00:20 +00:00
bjornv@webrtc.org
2111d3b0b0 Removed the vad_const files and added the constants to the files where they are 
used. Having them in a separate file did not add anything in readability or
conceptual overview.
Review URL: http://webrtc-codereview.appspot.com/230004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 12:58:34 +00:00
wu@webrtc.org
4ee906d297 When WEBRTC_VIDEO_ENGINE_FILE_API is not defined, disable the code in vie_file_impl.cc and vie_file_image.cc so that we can remove the libjpeg dependency. Also disable the auto test for the vie file api. 
Review URL: http://webrtc-codereview.appspot.com/227001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 17:56:38 +00:00
marpan@webrtc.org
5a3e20f678 Removed unused variables (build error) for test_fec. 
Review URL: http://webrtc-codereview.appspot.com/223001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 16:59:24 +00:00
pwestin@webrtc.org
1da1ce0da5 First implementation of simulcast, adds VP8 simulcast to video engine. 
Changed API to RTP module
Expanded Auto test with a test for simulcast
Made the video codec tests compile
Added the vp8_simulcast files to this cl
Added missing auto test file
Review URL: http://webrtc-codereview.appspot.com/188001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 15:19:55 +00:00
stefan@webrtc.org
4c059d87b3 Add metric for number of packets discarded by JB due to not being decodable 
Also fixes a couple of bugs related to sequence number wrap found while
testing.

BUG=
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Review URL: http://webrtc-codereview.appspot.com/218001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 07:35:37 +00:00
wu@webrtc.org
77d7d5455e Replace the DestroyDeviceInfo with a virtual destructor. 
Review URL: http://webrtc-codereview.appspot.com/212005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-12 16:57:53 +00:00
amyfong@webrtc.org
e5542a0af5 Add file record and play functions to voe_cmd_test, fix Play local file (path was incorrect) 
Fixed:
	24. Play local file (audio_long16.pcm) 
New:
	34. Record a PCM file 
	35. Play a previously recorded PCM file locally 
	36. Play a previously recorded PCM file as microphone 
Review URL: http://webrtc-codereview.appspot.com/209001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 20:30:56 +00:00
amyfong@webrtc.org
6330cf2a14 Fixed ViE AutoTest trace file names to be consistent 
Fixed some space issues in vie_autotest_custom_call.cc
Fixed incorrect default codec W&H for I420 in vie_autotest_custom_call.cc
Added functionality to modify a running custom call.  The following options were added:
0. Finished modifying custom call
1. Change Video Codec
2. Change Video Size by Common Resolutions
3. Change Video Size by Width & Height
4. Change Video Device
5. Record Incoming Call
6. Record Outgoing Call
7. Play File on Video Channel(Assumes you recorded incoming & outgoing call)
8. Print Call information

Tested with r670, builds fine on Ubuntu & Win7.  Mac is not building due to changes in r666, but this patch should be fine on top of it mac as well (compiles fine with r661).
Review URL: http://webrtc-codereview.appspot.com/188003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 18:17:22 +00:00
wu@webrtc.org
ea89922b56 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. 
BUG=
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Review URL: http://webrtc-codereview.appspot.com/213002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 17:13:51 +00:00
andrew@webrtc.org
199f4defd3 Rename all .cc files which include Objective-C headers to .mm. 
This allows the Mac Make build to pass. We were hacking it in XCode with "-x objective-c++", but gyp/Make doesn't seem to accept that flag.

Also switch Objective-C #includes to #imports.

There is one file missing from this: vie_autotest_main.cc, because it's required on multiple platforms. I'm not immediately sure what the best approach is there, but the Objective-C headers should be somehow hidden.
Review URL: http://webrtc-codereview.appspot.com/153005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 15:43:35 +00:00
henrike@webrtc.org
a0258defd4 Fixes test build errors (warnings treated as errors) in system_wrappers. 
Review URL: http://webrtc-codereview.appspot.com/212003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 14:49:27 +00:00
henrik.lundin@webrtc.org
26c9ff983e Add dummy implementation of DataLog::Combine method 
The dummy implementations of class methods are needed when
building without support for data logging (i.e., when
enable_data_logging != 1). The Combine method was missing
from data_log_dummy.cc.

Review URL: http://webrtc-codereview.appspot.com/220003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 14:43:41 +00:00
stefan@webrtc.org
791eec7424 Add API to get the number of packets discarded by the video jitter buffer due to being too late. 
BUG=
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Review URL: http://webrtc-codereview.appspot.com/200001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 07:53:43 +00:00
stefan@webrtc.org
06887aebae Fixes two bugs when decoding with packet losses. 
Disable _missingFrame bit since we can't set it correctly with FEC.

No longer return more than one decoded frame per Decode() call.
This is a work-around for a bug where the frame info map was popped more often than items were added to the map.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/215001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 14:17:46 +00:00
henrik.lundin@webrtc.org
1843664f2a DataLog: Changing from common_types to typedefs 
The file common_types.h cannot be used in data_log_c.h, since
the latter is a pure C header file, and common_types.h is
not. Changing to typedefs.h instead.

Review URL: http://webrtc-codereview.appspot.com/216001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 09:56:52 +00:00
tommi@webrtc.org
c0b2250b20 Fix the Windows build. 
Review URL: http://webrtc-codereview.appspot.com/213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 08:43:33 +00:00
henrik.lundin@webrtc.org
d855bd4d6f C wrapper for DataLog class 
A pure C wrapper for the DataLog class was created. Since templates
are not supported in C, the InsertCell method of the DataLog class
must be wrapped using one wrapper function for each data type. So far,
the wrapper includes int, float, double, Word32, UWord32, and Word64.

Unittests were created for the wrapper. A separate helper file was
included in the tests. This helper file was implemented as a C file,
in order to actually test the C linkage of the wrapper.
The unittests for DataLog were cloned to make versions that do the same
things but through the C wrapper interface. Restructured the code
so that the log file verification was not duplicated.

Review URL: http://webrtc-codereview.appspot.com/195003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 08:06:17 +00:00
tommi@webrtc.org
6364d128a1 Fix a couple of build warnings. 
Review URL: http://webrtc-codereview.appspot.com/214004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 08:04:59 +00:00
phoglund@webrtc.org
e95458c30a Started rewriting video_engine tests to use GUnit. 
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-10 07:23:51 +00:00
kjellander@webrtc.org
25e0b8e3a0 Python output flag and keyframe interval flags. 
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-07 07:52:00 +00:00
kjellander@webrtc.org
a31b254084 Python output flag and keyframe interval flags. 
Refactored main method into using 6 helper methods for better overview.

Review URL: http://webrtc-codereview.appspot.com/207001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-07 06:50:22 +00:00
mikhal@webrtc.org
80dd19be0a vplib tests: Removing old and unused file and directories. 
Note that the convert_test and scale_test directories are also removed. 
Review URL: http://webrtc-codereview.appspot.com/208001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 22:57:06 +00:00
henrike@webrtc.org
bf54ef9bb7 Removed code under a non-existing define. 
Review URL: http://webrtc-codereview.appspot.com/193006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 18:14:25 +00:00
henrike@webrtc.org
1a2933c71a Fixes a Valgrind warning triggering when the number of pending messages hit the limit. 
Review URL: http://webrtc-codereview.appspot.com/200002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 17:55:56 +00:00
andrew@webrtc.org
b2d4921f3b Remove trailing whitespace in AudioDevice. 
(That I introduced...)
Review URL: http://webrtc-codereview.appspot.com/198002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:34:36 +00:00
mikhal@webrtc.org
d6132f54d2 Review URL: http://webrtc-codereview.appspot.com/193007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:23:38 +00:00
kjellander@webrtc.org
35a1756502 First version of video quality measurement program and test framework. 
See https://docs.google.com/a/google.com/document/d/1w6Nrxw6yTg_sDu18Ux8oZPEMo5F_R-zt62udrmmTeOc/edit?hl=en_US
for background, details and additional instructions on usage.

BUG=
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Review URL: http://webrtc-codereview.appspot.com/175001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 06:44:54 +00:00
andrew@webrtc.org
3ce62fcfe4 Move merge_libs targets to their own gyp. 
The main reason is to depend on all ("*") targets in voice_engine.gyp and video_engine.gyp. We don't want the merge_lib targets building by default, since they do funny stuff like delete some libraries.
Review URL: http://webrtc-codereview.appspot.com/191003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 01:03:18 +00:00
kma@webrtc.org
af57de006a Some code style changes in audio_processing/ns/main/source/ by Astyle, 
with a little manual modification.
Review URL: http://webrtc-codereview.appspot.com/201002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 23:36:01 +00:00
henrik.lundin@webrtc.org
01ca01f6e6 Adding neteq_tests to modules tests 
Also moving neteq_tests.gyp and renaming to gypi. Cleaning up a
little in neteq_tests.gypi.

Review URL: http://webrtc-codereview.appspot.com/191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 20:38:19 +00:00
kma@webrtc.org
bbc1f10187 Changed modules/audio_processing/utility/Android.mk, to correct a build error in 
Android with the change from version r674.
Review URL: http://webrtc-codereview.appspot.com/197003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 18:09:02 +00:00
kma@webrtc.org
bf39ff4271 Some general optimization in NS. 
No big effort in introducing new style.
Speed improved ~2%.
Bit exact.
Will introduce mulpty-and-accumulate and sqrt_floor next, which increase speed another 2% or so.

Note: In function WebRtcNsx_DataAnalysis, did the block separation because I found one "if" case is more frequent than "else" within a for loop; rest is kind of code re-aligning.
Review URL: http://webrtc-codereview.appspot.com/181002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 17:10:06 +00:00
kma@webrtc.org
a58224f9f0 Introduced a SPL inline function (multiple-accumulate), for preformance in ARMv7. 
It's used in quite some occations over many modules.
Review URL: http://webrtc-codereview.appspot.com/178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 16:44:11 +00:00
stefan@webrtc.org
4b6f747373 Fixes a newly introduced bug in the jitter buffer where buffer reallocation 
causes corrupt pointers.
Review URL: http://webrtc-codereview.appspot.com/186003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:58:39 +00:00
stefan@webrtc.org
93d216c23f Fixed bug in jitter buffer which caused the missingFrames bit to never be set. 
Also updated the VP8 wrapper to return fully concealed frames (for rendering).

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/190003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:48:11 +00:00
stefan@webrtc.org
61b4abf1f8 Proper use of frame rate argument in generic_codec_test. 
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 06:40:21 +00:00
mikhal@webrtc.org
e06be4f678 video coding tests: Adding ssimFrame to interface 
Review URL: http://webrtc-codereview.appspot.com/188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@685 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:43 +00:00
mikhal@webrtc.org
ae7a0522c5 video_coding robustness: Updating hybrid mode's settings 
1. Disabling adjustment factor - temporary update. 
2. Enabling a windowed filtered loss for the hybrid mode.  
Review URL: http://webrtc-codereview.appspot.com/192003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:54:34 +00:00
marpan@google.com
f1f3fb33b5 Update to rate-mismatch factor in media_opt_util. 
Review URL: http://webrtc-codereview.appspot.com/193003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 19:09:45 +00:00
andrew@webrtc.org
f458916145 Returning errors if any of the Init() settings in VoE fail. 
There's no reason to try to continue if these simple settings fail; better to know about it immediately.

Also, readjusting the indentation to avoid breaking strings over several lines. This bends GStyle a bit, but it's well worth it to avoid the common "forgot to add a space" error.
Review URL: http://webrtc-codereview.appspot.com/173003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 15:22:28 +00:00
stefan@webrtc.org
5b91464edf Allow an aggregated partition to spill over to a new packet. 
Adds support for the case where the partition 0 and parts of partition 1
are transmitted in packet 1, and the end of partition 2 is transmitted
in packet 2.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/181003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 10:26:12 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC. 
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 08:18:10 +00:00
stefan@webrtc.org
f72c36763f Reverting changelist 666 since it broke the build on Mac. 
TBR=mflodman
Review URL: http://webrtc-codereview.appspot.com/187003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 07:37:41 +00:00
andrew@webrtc.org
6d169f2474 Fix Mac build error in vie_auto_test introduced in r666. 
COCOA_RENDERING was undefined. Committing without review.
Review URL: http://webrtc-codereview.appspot.com/191002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 06:00:42 +00:00
mflodman@webrtc.org
5eec6cf29a Started rewriting video_engine tests to use GUnit. 
- Added comments to the new test.
- Added a new mode to the vie_auto_test binary. It is now possible to pass --automated to it to make it run noninteractively. - To be precise, it will run everything that has been rewritten as GUnit tests, which currently is one "test suite" in the binary.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/168002
Patch from Patrik Hoglund <phoglund@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 12:24:13 +00:00
punyabrata@webrtc.org
6b6d08164f Remove assert "currentVoEMicLevel <= kMaxVolumeLevel". We ran into an issue on a Linux system where the currentVoEMicLevel was in fact greater than the kMaxVolumeLevel. Therefore we are removing this assert and capping the currentMicLevel to the maxVolumeLevel when this case is detected. 
Review URL: http://webrtc-codereview.appspot.com/180001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 17:45:03 +00:00
kma@google.com
c611b1a950 Bit-exact with non-Neon version. 
Review URL: http://webrtc-codereview.appspot.com/180002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 16:03:38 +00:00
bjornv@google.com
0beae6798d Removed level estimator calls, since it is not supported. There are still one place left; used within SetRTPAudioLevelIndicationStatus(). The error return value of level_estimator() has no effect there. 
The VoE auto tests have been updated as well.
Review URL: http://webrtc-codereview.appspot.com/178003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 14:08:19 +00:00
andrew@webrtc.org
18421f2063 Remove unnecessary include from NS interface. 
http://code.google.com/p/webrtc/issues/detail?id=46
Review URL: http://webrtc-codereview.appspot.com/183001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:50:52 +00:00
amyfong@webrtc.org
6a23ad5702 Fixed the CameraCap button to say Version, also change the function name inside ChannelDlg.cpp 
Review URL: http://webrtc-codereview.appspot.com/182001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 19:19:10 +00:00
amyfong@webrtc.org
2d08d43206 * Added modification of Start Bit Rate to vie_auto_test_custom_call 
* Added minor spacing and ":" for user input during vie_auto_test_custom_call
* Changed the default Video Port to 11111 and Audio Port to be 11113 to bring it inline with the WindowsTest application for ViE
Review URL: http://webrtc-codereview.appspot.com/181001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 17:46:45 +00:00
mikhal@webrtc.org
848fad23c6 video_coding: Updating media opt test - fixing call to protection callback. 
Review URL: http://webrtc-codereview.appspot.com/179003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 16:30:59 +00:00
xians@google.com
49d025f262 Get the right guid str for GetRecordingDeviceName 
Bug=http://code.google.com/p/webrtc/issues/detail?id=99
Test=none
Review URL: http://webrtc-codereview.appspot.com/183002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 14:43:06 +00:00
bjornv@google.com
a2c6ea09b0 Removed a segmentation fault error when processing near_file only. 
Review URL: http://webrtc-codereview.appspot.com/174001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 08:04:45 +00:00
kma@google.com
961885a8bb In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7. 
Review URL: http://webrtc-codereview.appspot.com/140010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-26 16:35:25 +00:00
mikhal@webrtc.org
e185e9f68a video_coding: updates to jitter buffer logic: Make sure that every frame is inserted only once to the list. 
Review URL: http://webrtc-codereview.appspot.com/165001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 22:02:40 +00:00
turajs@google.com
cf136186f5 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:49:25 +00:00
turajs@google.com
13335ccd7a Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:47:25 +00:00
turajs@google.com
610f478705 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:45:34 +00:00
turajs@google.com
53439d9982 Deleting matlab files 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 21:44:00 +00:00
amyfong@webrtc.org
713f91e12b Fixed vie_autotest_custom_call.cc minor issues. 
1. mirror of local render removed
2. the video device the user selected wasn't what was actually being used when the call is being made
3. fixed mentions of loopback calls
Review URL: http://webrtc-codereview.appspot.com/171001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:26 +00:00
mikhal@webrtc.org
105ff39dec video coding: updating offline tests. 
Additional clean-up to the offline test: Placing test callbacks in a designated file. 
Review URL: http://webrtc-codereview.appspot.com/167002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 16:41:11 +00:00
turajs@google.com
496ef8aca8 To fix warnings in test files. 
Review URL: http://webrtc-codereview.appspot.com/169001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 15:45:48 +00:00
bjornv@google.com
8e9e83b530 This CL adds guards against division by zero, that should fix http://b/issue?id=5278531 
In addition a read outside memory event has been detected and removed.
Also an improper noise weighting has been corrected.
Review URL: http://webrtc-codereview.appspot.com/152001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 12:39:47 +00:00
kjellander@webrtc.org
9e7774f163 Added compare methods for TickInterval class. 
This is useful to be able to sort them using the STL algorithm library.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/173002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 11:33:31 +00:00
bjornv@google.com
dc743a8bba Replaces a use of log2. 
I've replaced a log2 operation so it works on Windows.
Review URL: http://webrtc-codereview.appspot.com/171002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-23 08:13:53 +00:00
leozwang@google.com
90eff6c7c6 Fix compilation error in build-in AEC test 
Review URL: http://webrtc-codereview.appspot.com/164001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 18:02:03 +00:00
wu@webrtc.org
221b522118 Return the number of /dev/video* without trying to open it. 
Consider the case when there're /dev/video0 and /dev/video1. But for somereason the video0 is not in a correct state and can't be open. As a result, current NumberOfDevices will return 1, which is fine. However, we will then never be able to get the device we really want - /dev/video1. Consider the code below, the GetCaptureDevice will fail because it calls into DeviceInfoLinux::GetDeviceName(0, ...) which will again try to open the /dev/video0. So the root cause is the mismatching of the NumberOfDevices and GetDeviceName.

Since we will open the device in DeviceInfoLinux::GetDeviceName anyway, I think we should return the number of /dev/video* in DeviceInfoLinux::NumberOfDevices without trying to open it. Otherwise the DeviceInfoLinux::NumberOfDevices should return more information like which /dev/video* is valid which is not.

bool found = false;
for (int i = 0; i < vie_capture->NumberOfCaptureDevices(); ++i) {
  if (vie_capture->GetCaptureDevice(i, ...) == 0) {
    found = true;
    break;
  }
}
Review URL: http://webrtc-codereview.appspot.com/148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 16:57:15 +00:00
bjornv@google.com
65e6ab31eb Temporary log2 remove to build in chrome 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 11:56:46 +00:00
amyfong@webrtc.org
3be70ca17e Added mute, hold and typing detect to voe_cmd_test to increase functionality in the voe_cmd_test application. 
Typing Detect is applicable only for Mac.  
Review URL: http://webrtc-codereview.appspot.com/156002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@632 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 23:41:06 +00:00
wu@webrtc.org
a1930427af When WEBRTC_VIDEO_EXTERNAL_CAPTURE_AND_RENDER is defined, we should never try to use _ptrCaptureDeviceInfo. 
Review URL: http://webrtc-codereview.appspot.com/167001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@631 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 17:38:57 +00:00
leozwang@google.com
657f483c26 Fix compilation error 
Review URL: http://webrtc-codereview.appspot.com/162003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:20 +00:00
leozwang@google.com
ec5e87614e Enable OPENELSE defination when compile voice engine 
Review URL: http://webrtc-codereview.appspot.com/150005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 16:41:09 +00:00
pwestin@webrtc.org
741da942ec Added support for new RTCP message REMB (remote estimated max bitrate) 
Review URL: http://webrtc-codereview.appspot.com/149001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@628 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 13:52:04 +00:00
andrew@webrtc.org
86b85db67e Add missing intrinsic casts for VS 2005. 
Allows re-enabling SSE optimization on Windows.
Review URL: http://webrtc-codereview.appspot.com/161003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 18:48:25 +00:00
leozwang@google.com
522f42bb80 Add kPlatformAndroid to platform check function 
Review URL: http://webrtc-codereview.appspot.com/161002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 17:39:05 +00:00
andrew@webrtc.org
ed083d4079 Modify the _vadActivity member of the AudioFrame passed to AudioProcessing. 
This saves the user from having to explicitly check stream_has_voice(). It will allow typing detection to function, which relies on this behaviour.
Review URL: http://webrtc-codereview.appspot.com/144004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:28:51 +00:00
andrew@webrtc.org
94c7413b0d Allow echo metrics to be enabled in process_test. 
Review URL: http://webrtc-codereview.appspot.com/155002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 15:17:57 +00:00
henrik.lundin@webrtc.org
4c36d3b424 Fixing windows warnings in rtp_utility 
Adding explicit casting to bool to avoid warnings when compiling
in windows.

Review URL: http://webrtc-codereview.appspot.com/140002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 08:16:20 +00:00
andrew@webrtc.org
67812a4621 Temporarily disabling SSE2 on Windows again until we can build on VS 2005. 
Skipping review because the build is broken on Windows.
Review URL: http://webrtc-codereview.appspot.com/156003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 02:28:49 +00:00
andrew@webrtc.org
d02dc6e682 Removing bwe_standalone from modules.gyp 
Review URL: http://webrtc-codereview.appspot.com/144003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-17 00:44:23 +00:00
wjia@google.com
fdaee9c014 include build/common.gypi directly 
Review URL: http://webrtc-codereview.appspot.com/153006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-17 00:06:08 +00:00
andrew@webrtc.org
7b7c045b75 Fix MSVC issues in AEC to enable SSE2 optimization on Windows. 
Variables now declared at top of scope and replacing C casts with intrinsic cast functions.
Review URL: http://webrtc-codereview.appspot.com/160001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 22:51:57 +00:00
leozwang@google.com
b37ec71dbd Remove delay_estimator_float.c from android build 
Review URL: http://webrtc-codereview.appspot.com/161001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 21:50:36 +00:00
leozwang@google.com
a5700876c0 Add include path to auto test 
Review URL: http://webrtc-codereview.appspot.com/155001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 20:38:31 +00:00
andrew@webrtc.org
416d702ace Fix autotest error on non-Win platforms. 
Review URL: http://webrtc-codereview.appspot.com/149007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-16 14:22:39 +00:00
leozwang@google.com
ce95069ade Fix buidling error 
Review URL: http://webrtc-codereview.appspot.com/151002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 22:28:08 +00:00
wu@webrtc.org
fcd12b3b7d Add necessary spaces to log. 
Review URL: http://webrtc-codereview.appspot.com/148002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 20:49:50 +00:00
andrew@webrtc.org
4537c2a464 Remove the UNCONSTR code path from AEC. 
Leave the unconstrained filter adaptation in a commented out function. Consider using this for a low-complexity mode.
Review URL: http://webrtc-codereview.appspot.com/146001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 18:25:14 +00:00
tommi@webrtc.org
8dc3985a10 Fix windows build. 
Review URL: http://webrtc-codereview.appspot.com/150001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 15:01:04 +00:00
bjornv@google.com
b47d4b287d This CL includes a move of the fixed point delay estimator from aecm to apm/utility. There has also been a code change that makes it possible to enable/disable the far end alignment, so that we save complexity when used as a quality metrics. 
Review URL: http://webrtc-codereview.appspot.com/135014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 12:27:36 +00:00
henrik.lundin@webrtc.org
29fd9a5f30 Removing warnings in all NetEQ test targets 
Now all targets in neteq.gypi builds again. Also added payload type to
the log produced by RTPanalyze.

Review URL: http://webrtc-codereview.appspot.com/148001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-15 08:25:45 +00:00
andrew@webrtc.org
19eefdc9f0 Add a unit testing framework. 
Populate it with the beginnings of a resampler unit test to have it do someting.

Also fix a bug in resampler caught with the test ;)
Review URL: http://webrtc-codereview.appspot.com/135019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-14 17:02:44 +00:00
andrew@webrtc.org
b524f441d0 Correct some comment spelling errors. Skipping review. 
Review URL: http://webrtc-codereview.appspot.com/144002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 18:04:30 +00:00
andrew@webrtc.org
7a585a7903 Correct voe_auto_test file path on Windows. 
Needs to be changed due to the recent move of voice_engine.gyp.
Review URL: http://webrtc-codereview.appspot.com/144001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:46:36 +00:00
andrew@webrtc.org
a3c6d61c44 Integrate the built-in WASAPI AEC DMO to VoE. 
Review URL: http://webrtc-codereview.appspot.com/108006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:17:49 +00:00
leozwang@google.com
b1b3e67c97 Fix compilation errors 
Review URL: http://webrtc-codereview.appspot.com/142002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:16:24 +00:00
leozwang@google.com
803a5f2795 Add include path 
Review URL: http://webrtc-codereview.appspot.com/141003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:15:22 +00:00
andrew@webrtc.org
2cef36fa98 Fix Windows gyp run. 
On Windows, gyp seems to require valid source files. The matlab_plotting_test target was missing its one source file, so I removed the target.

Also moving bwe_standalone.gypi to the test include list.
Review URL: http://webrtc-codereview.appspot.com/143001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 17:03:15 +00:00
andrew@webrtc.org
f5fb095bf9 Fix audio processing tests gypi after recent changes. 
Review URL: http://webrtc-codereview.appspot.com/137025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-13 01:04:59 +00:00
andrew@webrtc.org
938a049dbe Clean up vie_auto_test.gypi. 
The most significant change is to use gyp's "copies" feature for copying the media files. This will now only trigger when needed, rather than with every build as before.
Review URL: http://webrtc-codereview.appspot.com/133020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 22:06:38 +00:00
marpan@google.com
45fa141f0a qm_select: changed default settings for uep. 
Review URL: http://webrtc-codereview.appspot.com/132015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 16:53:19 +00:00
henrik.lundin@webrtc.org
9f710d08e1 Switch to new sqrt in NetEQ 
Switched to WebRtcSpl_SqrtFloor instead of WebRtcSpl_Sqrt in
NetEQ. The output is not bit-exact, but subjective listening
tests show no audible difference. Analysis shows that almost
all of the difference is in changed delay.

The reference file for NetEQ's unit test was updated.

Review URL: http://webrtc-codereview.appspot.com/139019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 16:44:37 +00:00
kjellander@webrtc.org
f0a8464b74 Added more statistics during SSIM/PSNR calculation, including calculation of min/max value. 
Moved video_metrics.h into a GYP library so it can be used from other projects.

Review URL: http://webrtc-codereview.appspot.com/132013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 13:45:39 +00:00
xians@google.com
d3185fe219 refactor the gyp file to gypi file. 
Basically, the gypi file is a copy of gyp file, but has some difference on the
path of the dependencies.
Review URL: http://webrtc-codereview.appspot.com/137020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 12:24:39 +00:00
perkj@webrtc.org
0cc68dc38a Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface. 
Furthermore remove all static module creation and deletion functions.
Review URL: http://webrtc-codereview.appspot.com/133012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 08:53:36 +00:00
tina.legrand@webrtc.org
31c6b60456 Adding calls to Version functions for external codecs. 
Also clarified in comments where to put interface files for external codecs.
Review URL: http://webrtc-codereview.appspot.com/135017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 07:18:37 +00:00
zakkhoyt@google.com
b448ae229c Permanently adding additional logs 
Review URL: http://webrtc-codereview.appspot.com/137024

git-svn-id: http://webrtc.googlecode.com/svn/trunk@577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:41:49 +00:00
zakkhoyt@google.com
c6e8b72c83 Removing qualifiers on include path 
Review URL: http://webrtc-codereview.appspot.com/132014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:41:13 +00:00
marpan@google.com
30ecda146a media_opt_util: Added comment and lowered window size parameter. 
Review URL: http://webrtc-codereview.appspot.com/135018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 17:15:12 +00:00
marpan@google.com
3f28061f3a media_opt_util: Modification to correction factor in FEC overhead. 
Review URL: http://webrtc-codereview.appspot.com/133019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:39:40 +00:00
punyabrata@webrtc.org
955d0eed2f Removing echo warning because it seems to be flooding the logs 
anytime there is any echo. Secondly, this should be treated as
a warning in the sense that echo in the signal does not mean
something is wrong with the engine.f
Review URL: http://webrtc-codereview.appspot.com/139018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:28:51 +00:00
kma@google.com
0ada410d37 Corrected conditional compilations for spl inline functions in ARM. 
Review URL: http://webrtc-codereview.appspot.com/140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 16:23:50 +00:00
mikhal@webrtc.org
6f54c20703 video coding test: Adding MT functionality 
Review URL: http://webrtc-codereview.appspot.com/135008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 14:38:59 +00:00
henrik.lundin@webrtc.org
35dcc23110 Adding regression test to NetEQ 
The test inputs RTP packets from an RTPdump file into NetEQ
and compares the output to the corresponding reference file.
Test files are included.

The change also includes a new method in NETEQTEST_RTPpacket
class, which reads past the initial file header in an RTPdump
file.

Finally, a few warnings are removed.
Review URL: http://webrtc-codereview.appspot.com/138012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-09 08:01:16 +00:00
andrew@webrtc.org
b801571755 Comment out the ARCH_ARM define temporarily, to avoid duplicate definition conflict with the Android makefiles. 
Review URL: http://webrtc-codereview.appspot.com/135016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 18:37:59 +00:00
stefan@webrtc.org
06e2c11703 Remove unintentional printfs 
Review URL: http://webrtc-codereview.appspot.com/131018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 13:16:24 +00:00
stefan@webrtc.org
167328eab6 Disable libvpx partitions code for libvpx versions prior Cayuga. 
Necessary for WebRTC to build with Chromium. 
Also fixes the decoder wrapper's Reset() function so that properly
reinitializes the decoder.
Review URL: http://webrtc-codereview.appspot.com/132012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 13:05:48 +00:00
stefan@webrtc.org
9e812fca9f Adding missing parts related to VP8 partitions 
Review URL: http://webrtc-codereview.appspot.com/131017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 10:11:24 +00:00
stefan@webrtc.org
42ab82bf2f Disable independent partitions by default. 
Review URL: http://webrtc-codereview.appspot.com/140006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@559 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 06:55:29 +00:00
stefan@webrtc.org
c3d891059e Adds support for VP8 partitions 
This change adds support for VP8 partitions in the video jitter buffer and 
the VP8 encoder and decoder wrappers. The feature is currently disabled by
default since it requires a later version of libvpx.

With this change the jitter buffer will also start keeping track of each
packet header until decoding, and the VCMSessionInfo and VCMPacket objects 
will keep pointers into the encoded frame buffers.
Review URL: http://webrtc-codereview.appspot.com/137021

git-svn-id: http://webrtc.googlecode.com/svn/trunk@558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 06:50:28 +00:00
kma@google.com
b6fc9f1d6f optimized one spl function for AECM. 
AECM test results bit-exact.
Review URL: http://webrtc-codereview.appspot.com/139012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 17:20:49 +00:00
stefan@webrtc.org
3bbe41aad6 Add support for a numeric table id in DataLogs. 
Useful when logging in modules which have multiple instances and the logs
should be separated into different tables.
Review URL: http://webrtc-codereview.appspot.com/132003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:31:03 +00:00
henrik.lundin@webrtc.org
dd07d5932a Let VP8 decoder handle NULL codecSpecificInfo 
VP8Decoder::Decode() can now handle the case when
codecSpecificInfo is NULL. Previously, it would crash.

Review URL: http://webrtc-codereview.appspot.com/135015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:21:38 +00:00
henrik.lundin@webrtc.org
ea05973e68 Fixing VCM tests for VP8 
Removing asserts since the PictureID (and other parameters)
is now piped through codecSpecific. Also made sure the VCM
send callbacks (test code) copies the appropriate paramters.
Finally, enabling I420 in tests.

Review URL: http://webrtc-codereview.appspot.com/137022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:20:17 +00:00
henrika@google.com
73d65513f1 Adds reference counting to the ADM. 
This CL modifies the ADM interface to ensure that an external ADM
can't call Create and Destroy any longer.

It also contains some minor style nits to conform better with
the Chromium style guide.
Review URL: http://webrtc-codereview.appspot.com/133014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:11:18 +00:00
mflodman@webrtc.org
436289f622 Quick fix so ViE autotest doesn't terminate Linux windows twice on exit. 
Review URL: http://webrtc-codereview.appspot.com/131016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 08:41:57 +00:00
andrew@webrtc.org
b44172dab9 Fix "braces recommended" warning in audio_conference_mixer. 
Review URL: http://webrtc-codereview.appspot.com/131014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 18:04:32 +00:00
andrew@webrtc.org
f6478958a4 Fix "ignoring return value" in ViE tests. 
Review URL: http://webrtc-codereview.appspot.com/133016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 16:46:25 +00:00
mflodman@webrtc.org
c547ed99d4 Remove ViE auto test build warning. 
Review URL: http://webrtc-codereview.appspot.com/138018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 15:49:34 +00:00
mflodman@webrtc.org
b69bb56741 Remove 'using namespace' from ViE autotest files. 
Review URL: http://webrtc-codereview.appspot.com/138017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 14:12:07 +00:00
perkj@google.com
ac75cab618 Fix reference counting assert. 
Change assert("teo") to assert(!"teo") so that the assert is actually triggered.
Review URL: http://webrtc-codereview.appspot.com/133018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 13:58:34 +00:00
stefan@webrtc.org
269f8a14c6 Undoing change committed in r514 since it broke bandwidth estimation 
Review URL: http://webrtc-codereview.appspot.com/132011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@531 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 09:51:59 +00:00
andrew@webrtc.org
e0ed8b26de Fix "return value unused" warnings in voe_cmd_test release mode. 
Review URL: http://webrtc-codereview.appspot.com/140001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 02:36:17 +00:00
andrew@webrtc.org
b01a62385f Fix "braces recommended" warning in video_engine. 
Review URL: http://webrtc-codereview.appspot.com/137019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-06 02:30:54 +00:00
xians@google.com
b875349537 fixing a bug in GetPlayoutDeviceName, previously it returns name as guid. 
Bug=http://code.google.com/p/webrtc/issues/detail?id=77
Test=none
Review URL: http://webrtc-codereview.appspot.com/135011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@528 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-05 12:17:30 +00:00
perkj@google.com
ea72c34fb9 Temporary add dummy implementation to RefCountModule. The reason is so that ADM and VideoCapture implementations can change to refcounted versions before forcing them. 
Review URL: http://webrtc-codereview.appspot.com/139014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-05 11:11:04 +00:00
henrik.lundin@webrtc.org
1e53166569 Fix VP8 tests 
These are changes that make the VP8 tests work again after the
wrapper was updated. The codec specific info is now propagated
properly through the encoder callback and into the queue struct.

Also added an fclose to get rid of a valgrind warning.
Review URL: http://webrtc-codereview.appspot.com/138011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@526 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-05 07:23:33 +00:00
leozwang@google.com
9d23ba096d Make test app work on android 
Review URL: http://webrtc-codereview.appspot.com/137014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-04 03:33:44 +00:00
zakkhoyt@google.com
fb298d3783 Modified path on speex lib 
Review URL: http://webrtc-codereview.appspot.com/137018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 22:06:49 +00:00
andrew@webrtc.org
413b993166 Put some table size information in one place. 
Motivated by fixing an unused variable warning in release mode.
Review URL: http://webrtc-codereview.appspot.com/132007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 22:03:56 +00:00
turajs@google.com
d7a41774ce header included twice. 
Review URL: http://webrtc-codereview.appspot.com/139013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 20:52:47 +00:00
perkj@google.com
88a0da8fde Add ref_count.h to gyp file. 
Review URL: http://webrtc-codereview.appspot.com/133013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:51:35 +00:00
perkj@google.com
9de5917776 Add an implementation of reference count to webrtc. 
Used for instantiating objects of RefCountModule.
Review URL: http://webrtc-codereview.appspot.com/135009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:24:51 +00:00
henrik.lundin@webrtc.org
2641fd1d19 Remove warnings in vp8_test 
Most modifications are either reordering of the initializers in constructors, removed unused variables, or comparison mismatches taken care of. A few other special cases are commented.
Review URL: http://webrtc-codereview.appspot.com/132008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 12:09:07 +00:00
perkj@google.com
ef04cf4b2e Adding reference counted version of the module interface. 
The reason for this is that we would like to have reference counting on the modules you can register externally with ViE and VoE.
Currently we plan to use this on the ADM, VideoCapture module and VideoRenderModule.
Review URL: http://webrtc-codereview.appspot.com/138010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-02 09:47:28 +00:00
andrew@webrtc.org
4d905f88c6 Fix clang warnings in rtp. 
Review URL: http://webrtc-codereview.appspot.com/132006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 19:22:27 +00:00
andrew@webrtc.org
f1f93d822e Remove warning settings more stringent than Chromium's common.gypi. 
Review URL: http://webrtc-codereview.appspot.com/131012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:57:44 +00:00
andrew@webrtc.org
a80d026517 Fix clang warnings in voice engine. 
Review URL: http://webrtc-codereview.appspot.com/133008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:30:09 +00:00
andrew@webrtc.org
bbd8908664 Fix clang warnings in video coding. 
Review URL: http://webrtc-codereview.appspot.com/138007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 17:30:01 +00:00
tina.legrand@webrtc.org
84519ec0a2 Fixing some inconsistencies in WebRTC audio coding module. I've added setup information for all codecs which are not part of WebRTC, but possible to hook in. 
Please help me review.
Henrik: review neteq_defines.h
Turaj: review all files, but the one Henrik reviews.
Zakk: FYI only.
Review URL: http://webrtc-codereview.appspot.com/138004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 07:47:31 +00:00
zakkhoyt@google.com
d9e11b429e Review URL: http://webrtc-codereview.appspot.com/137004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 00:54:32 +00:00
andrew@webrtc.org
777ef59394 Fix clang warnings in video engine. 
There are a number of namespace related warnings remaining in the video engine tests.
Review URL: http://webrtc-codereview.appspot.com/135007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 00:41:31 +00:00
marpan@google.com
243db12616 media_opt_util: Fixed an assert and some code cleanup for AvgRecoveryFEC function. 
Review URL: http://webrtc-codereview.appspot.com/139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:52 +00:00
wu@webrtc.org
b15bfd32d7 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. 
* Fix one issue in webrtcvideoengine where we should remove the renderer before adding a new one.
Review URL: http://webrtc-codereview.appspot.com/137011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:44 +00:00
turajs@google.com
ebb2744337 To fix warning for unused variable. And fix some warning in test. 
Review URL: http://webrtc-codereview.appspot.com/131010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 21:28:08 +00:00
turajs@google.com
eaf3185105 Take care of unused variable. 
Review URL: http://webrtc-codereview.appspot.com/137013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 21:27:53 +00:00
andrew@webrtc.org
9562a3664c Last fixes to build with gcc 4.6. 
Set but unused parameter/variable warnings.
http://code.google.com/p/webrtc/issues/detail?id=52
Review URL: http://webrtc-codereview.appspot.com/139006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 18:50:12 +00:00
andrew@webrtc.org
830099eba4 Add a gyp flag to disable video functionality from dependencies shared by voice and video engine. 
Currently, this is just the utility module. It relies on the already existing WEBRTC_MODULE_UTILITY_VIDEO define.
Review URL: http://webrtc-codereview.appspot.com/133007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 17:03:54 +00:00
pwestin@webrtc.org
e9f0e2eb20 Moved _rtpReceiver to protected 
Review URL: http://webrtc-codereview.appspot.com/132005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 13:16:52 +00:00
tommi@webrtc.org
c7d5f6249b Fix build errors on Windows. 
Since this is a C file, variables must be declared at the top of the function
so I'm moving the fix for the warning (inst = NULL) to the bottom of the funciton.
Otherwise, the compiler will complain when it sees int i; on systems that do
not have WEBRTC_BIG_ENDIAN defined.
Review URL: http://webrtc-codereview.appspot.com/139005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 12:11:24 +00:00
turajs@google.com
74c640aebb fix build break 
Review URL: http://webrtc-codereview.appspot.com/132004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:44:24 +00:00
turajs@google.com
7796c02b42 Wrap encode, decode, PLC NB functions in #define to avoid warnings. 
Review URL: http://webrtc-codereview.appspot.com/133005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:30:17 +00:00
turajs@google.com
8ecd0e8f3d Remove Clang warning for PCM16B. 
Review URL: http://webrtc-codereview.appspot.com/137006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 20:29:50 +00:00
punyabrata@google.com
eba8c32840 Resolving a race condition issue related to using shared devices 
(e.g. usb headsets) where we were not stopped the shared callback
until both StopPlayout() and StopRecording() are called. Google
internal bugid 4478351
Review URL: http://webrtc-codereview.appspot.com/130001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 14:32:22 +00:00
xians@google.com
3266d8d85d have the voe_cmd_test compiled with external transport enabled. 
Bug=http://code.google.com/p/webrtc/issues/detail?id=43
Test=none
Review URL: http://webrtc-codereview.appspot.com/133006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 08:29:07 +00:00
xians@google.com
e74a9ea303 AudioDeviceUtility::WaitForKey() pulls two characters if the first one is a newline, but discards the final value. 
The current code assigns that second value to a local variable, which generates a set-but-unused warning on gcc 4.6.0. Instead, cast the result away.

I also refactor the code a bit by adding the right indentation and removing empty lines.

Bug=http://code.google.com/p/webrtc/issues/detail?id=53
Test=none
Review URL: http://webrtc-codereview.appspot.com/135005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 08:27:02 +00:00
xians@google.com
932096c84f Porting gtalk alsa impl from depot to webrtc 
Review URL: http://webrtc-codereview.appspot.com/123002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 07:41:55 +00:00
mikhal@webrtc.org
46171cf546 video coding tests: Adding a Normal distribution to simulate packet arrival times 
Review URL: http://webrtc-codereview.appspot.com/138003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 23:38:04 +00:00
henrik.lundin@webrtc.org
8571af7be6 Updating to new VP8 rtp format 
The VP8 packetizer and tests have been updated to the new
RTP draft (http://tools.ietf.org/html/draft-ietf-payload-vp8-01).
The receive-side parser is also updated, and a new unit test
is implemented for it. Finally, some data traversing work to
get the parsed information into the decoder.
Review URL: http://webrtc-codereview.appspot.com/116011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 15:37:12 +00:00
hellner@google.com
09734086c6 Fixes build issue in http://code.google.com/p/webrtc/issues/detail?id=56. 
Review URL: http://webrtc-codereview.appspot.com/131008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 14:10:01 +00:00
tina.legrand@webrtc.org
81fd2bfbba New ACM codec database, created at compile time. 
Review URL: http://webrtc-codereview.appspot.com/127002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 11:18:44 +00:00
tina.legrand@webrtc.org
a41b4ce7da Changing iLBC to use the new improved SQRT, WebRtcSpl_SqrtFloor(). 
The bit-stream has not change with the new SQRT, but the output signal has. The change in output is small, and all test-files pass a subjective quality test.
New test-files will be committed to svn after this CL.
Review URL: http://webrtc-codereview.appspot.com/136001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 08:19:30 +00:00
stefan@webrtc.org
c9cff24ff0 Adding classes to be used for logging data within the engines and the 
components for offline processing. Data logged with these classes can
conveniently be parsed and processed with e.g. Matlab.
Review URL: http://webrtc-codereview.appspot.com/95009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:39:02 +00:00
xians@google.com
c9b75e0a4b removing the warnings from the voe tests. 
Bug=http://code.google.com/p/webrtc/issues/detail?id=61
Test=None
Review URL: http://webrtc-codereview.appspot.com/139003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:30:16 +00:00
tina.legrand@webrtc.org
2aa5d500af Issue reported in WebRTC. A variable is defined and set, but never used. 
Review URL: http://webrtc-codereview.appspot.com/139001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 06:36:37 +00:00
henrik.lundin@webrtc.org
36450af2b3 Removing unsupported codecs from ptypes file 
The file ptypes.txt tells test program NetEqRTPplay how to
map the RTP payload types in an RTP file. Now removing payload
types that are not supported in WebRTC.
Review URL: http://webrtc-codereview.appspot.com/119009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-27 01:25:35 +00:00
andrew@webrtc.org
bd4494cb20 Remove the divide-by-2 when mixing. 
Review URL: http://webrtc-codereview.appspot.com/137007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 22:58:00 +00:00
mikhal@webrtc.org
b7ac56d92b video coding tests: updating quality tests following r466 
Review URL: http://webrtc-codereview.appspot.com/131009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:35 +00:00
mikhal@webrtc.org
d24a97fae1 video coding test: deleting unused file(resampler_test.cc) 
Review URL: http://webrtc-codereview.appspot.com/137008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:17 +00:00
mikhal@webrtc.org
2c3b1fb4f3 video_coding tests: removing unused functionality from test_util 
Review URL: http://webrtc-codereview.appspot.com/137009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:18:04 +00:00
mikhal@webrtc.org
a057a9561c video_coding: Updating protection logic in media optimization utility: 
1. Changing protection logic structure: Accepts only one method (not a list)
2. Removed unused code (unreferenced protection methods)
3. Removed inline constructors/destructors.  
Review URL: http://webrtc-codereview.appspot.com/120005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 21:17:34 +00:00
mikhal@webrtc.org
552f173979 video_coding: Moving video metrics computation to a designated file. 
This is the first stage of a general clean-up to test_util. Will try to divide this clean-up to small changes, so it will be easier to review. 
Review URL: http://webrtc-codereview.appspot.com/120004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:38:09 +00:00
andrew@webrtc.org
e46d69f762 Fix gcc 4.6 set but unused warnings in AEC. 
Review URL: http://webrtc-codereview.appspot.com/134003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:20:54 +00:00
andrew@webrtc.org
ffbe7a75fd Cast away the unused state argument value to silence gcc 4.6 warnings. 
The WebRTC C wrapper for the G711 codec doesn't actually use the 'state' 
argument, but declares one anyway for API uniformity.

At the beginning of functions like WebRTCG711_EncodeA(), there's a stanza:

    // Set to avoid getting warnings
    state = NULL;

This might work around an unused parameter warning, but under gcc 4.6.0 
it ends up generating another warning, that state is set but not used.  

Casting the assignment to void silences the warning, restoring 
compilation under -Werror.

Reported as https://code.google.com/p/webrtc/issues/detail?id=50
Review URL: http://webrtc-codereview.appspot.com/135002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:16:30 +00:00
turajs@google.com
7f2bbbbefd To remove all calls involving scratch-memory 
Review URL: http://webrtc-codereview.appspot.com/129001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 16:03:49 +00:00
turajs@google.com
ac55f7b33c Review URL: http://webrtc-codereview.appspot.com/115004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 16:02:16 +00:00
xians@google.com
7659b366ac revert the file path in the voe_auto_test 
Review URL: http://webrtc-codereview.appspot.com/131007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 14:13:27 +00:00
xians@webrtc.org
c57f9c38ad Using IAudioEndpointVolume in IsSpeakerMuteAvailable and IsMicrophoneMuteAvailable to be consistent with SpeakerMute and MicrophoneMute APIs. 
Review URL: http://webrtc-codereview.appspot.com/112007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 12:28:33 +00:00
mflodman@webrtc.org
4fcb0caf78 Removing warning in video capture module for linux and auto test. 
Review URL: http://webrtc-codereview.appspot.com/134002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 10:54:48 +00:00
hellner@google.com
23a8065e36 Fixed broken build due to r453. 
Review URL: http://webrtc-codereview.appspot.com/131004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 21:40:11 +00:00
zakkhoyt@google.com
59af6f1434 Porting Mac keypress detection from GIPS repository. 
Mac keypress detection was added specifically for GTalk.
Review URL: http://webrtc-codereview.appspot.com/124001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 20:30:25 +00:00
mikhal@webrtc.org
ba9bd692ea video_coding_tests: Fix build error 
Review URL: http://webrtc-codereview.appspot.com/132001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 20:12:03 +00:00
stefan@webrtc.org
44d356d6df Fix unused variable warning in spatial_resampler.cc 
Issue 60: [Patch] Fix unused variable warning in spatial_resampler.cc
Review URL: http://webrtc-codereview.appspot.com/125003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 07:53:53 +00:00
turajs@google.com
5cc9c68e8d Fixing a warning discovered while compiling with clang. 
Review URL: http://webrtc-codereview.appspot.com/120003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 21:20:33 +00:00
marpan@google.com
057efc8f98 Removed unused variables and unnecessary assert: causing build error in vpm_test. 
Review URL: http://webrtc-codereview.appspot.com/128001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 20:53:15 +00:00
andrew@webrtc.org
4f390000dd Fix warnings on Ubuntu 11.04 (gcc 4.5) 
http://code.google.com/p/webrtc/issues/detail?id=63
Review URL: http://webrtc-codereview.appspot.com/125004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 20:35:35 +00:00
wu@webrtc.org
37fd004c69 Remove the X11 headers we don't need. 
Review URL: http://webrtc-codereview.appspot.com/123003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 17:06:38 +00:00
frkoenig@google.com
cf36b2aa9b Match new[] / delete [] 
Quiet valgrind warnings by correctly matching 
new[] with delete[].
Review URL: http://webrtc-codereview.appspot.com/126005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 15:48:47 +00:00
stefan@webrtc.org
49cbc512ae Fix unused variable warning in video_coding. 
Issue 57: [Patch] Fix unused variable warnings in the video_coding module
Review URL: http://webrtc-codereview.appspot.com/126004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 08:51:08 +00:00
andrew@webrtc.org
7f593c1e62 Fix gcc 4.6 unused variable warnings in audio_processing. 
Issues:
http://code.google.com/p/webrtc/issues/detail?id=54
http://code.google.com/p/webrtc/issues/detail?id=55
Review URL: http://webrtc-codereview.appspot.com/121003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 01:00:50 +00:00
mikhal@webrtc.org
6724cf8183 VP8: Adding a flag to indicate the libvpx version. When in Cayuga, additional API's will be used. 
Review URL: http://webrtc-codereview.appspot.com/120006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 00:51:36 +00:00
wjia@google.com
4482b04207 revert r430 to keep webrtc always ready to roll in chromium. 
r430 will be used when libvpx in chromium is rolled to Cayuga.
Review URL: http://webrtc-codereview.appspot.com/119008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 23:41:00 +00:00
wjia@google.com
f9f1deba8f Get ready for libvpx Cayuga (v0.9.7-p1). 
When building with Chromium, on Windows, only header files are needed; otherwise, libvpx.gyp:libvpx is needed.

This patch is based on http://webrtc-codereview.appspot.com/91019/
Review URL: http://webrtc-codereview.appspot.com/122005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 23:08:30 +00:00
hellner@google.com
a386fc0a8b Fixes build warnings due to unused variables. 
Code directly from http://code.google.com/p/webrtc/issues/detail?id=58.
Review URL: http://webrtc-codereview.appspot.com/119007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 21:26:09 +00:00
hellner@google.com
9aa9996a19 Different solution than the one suggested in http://code.google.com/p/webrtc/issues/detail?id=56 however, should solve the same problem. 
Review URL: http://webrtc-codereview.appspot.com/126003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@427 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 21:25:55 +00:00
wu@webrtc.org
ae53bf87d7 The variable ‘dummy’ set but not used being treated as errors in Fedora. 
Review URL: http://webrtc-codereview.appspot.com/122004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 18:56:56 +00:00
andrew@webrtc.org
ceb148ce59 Fix compile warnings in Release configuration. 
Review URL: http://webrtc-codereview.appspot.com/119003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@424 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 17:53:54 +00:00
mikhal@webrtc.org
06ad81fd58 video_coding: changing the UpdateMethod function (protection settings). 
Review URL: http://webrtc-codereview.appspot.com/126002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@423 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 16:03:59 +00:00
perkj@google.com
12f1fc4fe5 Fix initialization defect in constructor webrtc::ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(WebRtc_Word32, bool) initialization list. 
Review URL: http://webrtc-codereview.appspot.com/125002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 14:26:33 +00:00
perkj@google.com
a3fc1aac52 Fix Issue 59. Fix a constructor cast warning in video_X11_channel.cc. 
http://code.google.com/p/webrtc/issues/detail?id=59.
Review URL: http://webrtc-codereview.appspot.com/122002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 13:38:14 +00:00
pwestin@webrtc.org
a070adbab2 Moved member RTPSender from private to protected. 
Review URL: http://webrtc-codereview.appspot.com/119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@420 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 11:17:03 +00:00
xians@webrtc.org
9d64705dea The method AudioDeviceWindowsWave::RecProc can use uninitialized variables t1 and t2. 
This results in an exception in debug mode and unpredictable behavior in release mode.

The problem is at line 3514:

        // Calculate processing time
        consumedTime = (int)(t2.QuadPart-t1.QuadPart);

Bug=http://code.google.com/p/webrtc/issues/detail?id=42
Test=none
Review URL: http://webrtc-codereview.appspot.com/121002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@419 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 09:14:56 +00:00
punyabrata@webrtc.org
5895ea1573 Fixes volume problem controls, happening with some Logitech headsets. Originally submitted as gips p4 depot CL 38122. 
Review URL: http://webrtc-codereview.appspot.com/116008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 22:46:38 +00:00
punyabrata@google.com
9695e75fbd Resolve a crash related to pulseAudio where we need to check if 
pa_context_get_source_info_by_name/pa_context_get_sink_info_by_name 
has early failure and returns NULL,then to avoid WaitForOperationCompletion
from crashing, paOperation must be checked to ensure it is not NULL.

Please note that ::WaitForOperationCompletion( ) is a void function
therefore returning without any error code if paOperation is indeed NULL

Google internal bug id [4356730]
Review URL: http://webrtc-codereview.appspot.com/122003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 22:35:14 +00:00
cduvivier@google.com
288c8698cb Optimization of 'cftmdl': 
* scalar optimization, vectorization.
* 1.7% AEC overall speedup for the straight C path.
* 9.2% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/109008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 21:55:33 +00:00
mikhal@webrtc.org
0e16b78ec9 VP8: Removing VP8Latest flag following the update to the Cayuga release 
Review URL: http://webrtc-codereview.appspot.com/116012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 21:25:39 +00:00
mikhal@webrtc.org
d70b77d19f video_coding: Updating offline tests 
1. Adding parameters to the  input arguments. 
2. Introducing delay in the RTPSend callback
3. Clean-up
Review URL: http://webrtc-codereview.appspot.com/116006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 21:08:15 +00:00
tommi@webrtc.org
adb23827c1 Fix windows build. 
TBR=andrew
Review URL: http://webrtc-codereview.appspot.com/125001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-20 13:43:44 +00:00
andrew@webrtc.org
f81f9f8c2a Add -Werror and -Wextra to the Linux build. 
Includes all fixes required for -Wextra.
Review URL: http://webrtc-codereview.appspot.com/117006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 22:56:22 +00:00
frkoenig@google.com
9139fddf0e Optimize ssim_8x8 for SSE2. 
Code was pulled from libvpx assembly and converted
to intrinsics.
Review URL: http://webrtc-codereview.appspot.com/122001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 22:33:08 +00:00
andrew@webrtc.org
6423509efc Fix Valgrind warnings in audio_processing. 
Review URL: http://webrtc-codereview.appspot.com/119001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 21:22:08 +00:00
frkoenig@google.com
f53055d60e Use optimized SSIM from libvpx. 
Review URL: http://webrtc-codereview.appspot.com/117005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 18:16:59 +00:00
leozwang@google.com
412889a2a9 Some cleanup in test app 
This CL is to keep track of work and demonstrate a way to do ndk build in case it's needed. But ndk-build doesn't work yet because of many reasons, issues will be addressed in future if it's needed.
Some minor changes in source files to make them pass compiler.
Review URL: http://webrtc-codereview.appspot.com/107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 15:34:34 +00:00
xians@google.com
54e4691e20 change from ./test/data/voice_engine/audio_long16.pcm to ../../test/data/voice_engine/audio_long16.pcm 
Review URL: http://webrtc-codereview.appspot.com/115003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 08:22:14 +00:00
holmer@google.com
2cdc7b94d2 Enabling libvpx error concealment. 
Review URL: http://webrtc-codereview.appspot.com/116010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-18 09:35:45 +00:00
tina.legrand@webrtc.org
2e09692006 This CL fixes the warnings when building acm or acm test. 
Review URL: http://webrtc-codereview.appspot.com/109013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-18 06:20:30 +00:00
frkoenig@google.com
ff476c6c4e Optimize temporal difference metric for SSE2. 
Quick optimizations using intrinsics.
Review URL: http://webrtc-codereview.appspot.com/115006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 23:24:57 +00:00
mikhal@webrtc.org
685383dd37 video_coding/media_opt_util: Removing windows warnings 
Review URL: http://webrtc-codereview.appspot.com/113006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 23:19:54 +00:00
frkoenig@google.com
31f24de5e3 Use the correct include file. 
There shouldn't be a dependency on spatial_resampler.h, defines
that were needed are in video_processing_defines.h.
Review URL: http://webrtc-codereview.appspot.com/117003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 22:23:28 +00:00
turaj@webrtc.org
ca758c268e To remove warnings found by clang 
Review URL: http://webrtc-codereview.appspot.com/113005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 20:25:36 +00:00
mikhal@webrtc.org
ebeb5a656b video_coding - JB: Ensuring that every frame is inserted only once to the list 
Review URL: http://webrtc-codereview.appspot.com/114006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 18:43:21 +00:00
henrik.lundin@webrtc.org
473bac8582 Propagate codec specific info to decoder 
Add explicit use of CodecSpecificInfo to VCMGenericDecoder and
the codecs (VP8 and I420). Propagate information from
WebRtcRTPHeader in VCM (IncomingPacket) to GenericDecoder.
Review URL: http://webrtc-codereview.appspot.com/109011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 09:47:33 +00:00
punyabrata@webrtc.org
7049389794 Fixes volume problem controls, happening with some Logitech headset [from gips P4 depot CL 38122] 
Review URL: http://webrtc-codereview.appspot.com/108004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 00:15:52 +00:00
punyabrata@webrtc.org
7bd857bc99 Apple Airfoil audio casting software crash resolution [Google Internal buglist id: 4444985] 
Review URL: http://webrtc-codereview.appspot.com/109001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 00:01:05 +00:00
hellner@google.com
aab69a4145 removed trailing whitespaces. 
Review URL: http://webrtc-codereview.appspot.com/109010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 23:17:32 +00:00
marpan@google.com
771ca422df Fixed assert error in media_opt_util that may have caused index for look-up table to be out of range. 
Review URL: http://webrtc-codereview.appspot.com/112005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 20:51:04 +00:00
kma@google.com
78dc99e2a1 performance in ARM V7 improved 7~8 %. 
Review URL: http://webrtc-codereview.appspot.com/114007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 20:00:18 +00:00
hellner@google.com
977c2966fc Review URL: http://webrtc-codereview.appspot.com/109006 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 17:30:30 +00:00
andrew@webrtc.org
ad2a37093a Fix unused variable warnings. 
Review URL: http://webrtc-codereview.appspot.com/114005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 17:18:17 +00:00
andrew@webrtc.org
5cdec9e399 Fix common_audio warnings. 
- Unused variable.
- Explicit operator precedence.
Review URL: http://webrtc-codereview.appspot.com/112004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 16:39:32 +00:00
leozwang@google.com
7189270640 Fix compliation error 
Review URL: http://webrtc-codereview.appspot.com/114003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 16:37:45 +00:00
hellner@google.com
17a018716c Minimal fixes to remove build warnings. 
Review URL: http://webrtc-codereview.appspot.com/109007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 16:23:46 +00:00
hellner@google.com
1abe7a0555 Minimal fixes to remove build warnings. 
Review URL: http://webrtc-codereview.appspot.com/116004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 15:52:57 +00:00
hellner@google.com
064a8dfaa2 Minimal fixes to remove build warnings. 
Review URL: http://webrtc-codereview.appspot.com/113004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 15:52:28 +00:00
xians@google.com
0134facd53 Get rid of the ADM test warnings. 
Review URL: http://webrtc-codereview.appspot.com/116001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 09:30:51 +00:00
mflodman@webrtc.org
e66a0c0ce7 Correcting one bug and changing sleep time while waiting for packets. 
It seems like Linux might need more than one second to receive first packets, will dig into this later. Correcting/changing logs.
Review URL: http://webrtc-codereview.appspot.com/114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 08:11:15 +00:00
xians@google.com
bf5d2ba8fb This CL will exclude the built-in ADM when building with chromium. 
When building with chromium, we use dummy ADM for all the platforms.
Review URL: http://webrtc-codereview.appspot.com/108001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 07:44:19 +00:00
kma@google.com
8dd7466b52 2nd check in 
Review URL: http://webrtc-codereview.appspot.com/112002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 03:28:28 +00:00
wu@webrtc.org
3d48d5b107 Bug fix. When GetNumCores failed it will return -1, which is a huge number if cast to unsigned int. 
Review URL: http://webrtc-codereview.appspot.com/116005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 00:03:36 +00:00
hellner@google.com
56294c61c9 Review URL: http://webrtc-codereview.appspot.com/114002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 22:47:40 +00:00
andrew@webrtc.org
3296d76c50 Remove unneeded G167 define from AEC. 
Review URL: http://webrtc-codereview.appspot.com/114001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 21:40:38 +00:00
frkoenig@google.com
241c4d0e8d Remove numPixels. 
The number of pixels was used in both the numerator and denominator of
the final calculation and is therefore not needed.
Review URL: http://webrtc-codereview.appspot.com/112003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 19:08:48 +00:00
leozwang@google.com
95222074ba Some changes made in android makefile 
Review URL: http://webrtc-codereview.appspot.com/109003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 17:40:56 +00:00
frkoenig@google.com
6d171c4f98 SSE2 optimizations for spatial content analysis. 
Using intrinsics to work on 16 pixels at a time when
computing the spatial metrics.
Review URL: http://webrtc-codereview.appspot.com/92013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 15:56:23 +00:00
mflodman@webrtc.org
3cbbcd98b6 Removing Linux warnings for auto test. 
Review URL: http://webrtc-codereview.appspot.com/116002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 15:50:59 +00:00
xians@google.com
3ab5d5f4d0 This patch just gets rid of the rest of warnings on voe_auto_test and adm. 
Trivial changes.
Review URL: http://webrtc-codereview.appspot.com/111001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 11:07:54 +00:00
holmer@google.com
155188ce40 Handle all VP8 packets within a frame as depending on the previous packet 
This is temporary until the VP8 receiver support fragments.
Review URL: http://webrtc-codereview.appspot.com/113002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-15 09:21:27 +00:00
andrew@webrtc.org
d11aa2b215 Fix build error on Windows due to gcc specific attribute. 
Review URL: http://webrtc-codereview.appspot.com/115001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-14 23:25:28 +00:00
andrew@webrtc.org
2be2f694a4 Remove implicit declaration warnings. 
Review URL: http://webrtc-codereview.appspot.com/113001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-13 18:22:42 +00:00
kma@google.com
1959e6fcb5 1st check-in for AECM Neon optimization. 
Review URL: http://webrtc-codereview.appspot.com/104001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-13 06:33:38 +00:00
leozwang@google.com
4033e1245d Change folder name from Android to android 
Review URL: http://webrtc-codereview.appspot.com/106002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-12 15:54:48 +00:00
leozwang@google.com
c5597353c6 Change folder name from Android to android 
Review URL: http://webrtc-codereview.appspot.com/107001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-12 15:53:44 +00:00
marpan@google.com
319fbd107d Added simple model to test_util.cc to simulate bursty packet loss. 
Review URL: http://webrtc-codereview.appspot.com/104002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 22:51:58 +00:00
mikhal@google.com
60873adc3e rtp_sender_video: Modify behavior on send video packet error. This issue was already updated in CL r217, and accidentally reverted in CL r231. 
Review URL: http://webrtc-codereview.appspot.com/106004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 22:30:00 +00:00
mallinath@google.com
16f1dfc953 Review URL: http://webrtc-codereview.appspot.com/92006 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 18:22:17 +00:00
leozwang@google.com
911e36c525 Also change folder name in audio_device test 
Review URL: http://webrtc-codereview.appspot.com/105002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 17:04:26 +00:00
leozwang@google.com
a9b48c3b64 Change test/Android to test/android 
Review URL: http://webrtc-codereview.appspot.com/106003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 16:33:10 +00:00
xians@google.com
0fe2627e6f adding audio_device_utility_mac.cc, which was misssing from previous CL. 
Review URL: http://webrtc-codereview.appspot.com/106001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 14:32:40 +00:00
xians@google.com
77f66051e8 delete old folders 
Review URL: http://webrtc-codereview.appspot.com/105001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 13:58:34 +00:00
xians@google.com
68efa21abc This CL refactor the audio_device.gyp to make use of the chromiurm _os suffix filter. 
This CL looks big, but most of the changes are only renaming. I hope it is all right for you.

Changes include:
rename source/Windows to source/win
rename source/Linux to linux
rename source/Mac to source/mac

Also rename some files to  follow chromium standard, for examples:
from alsasymboltable.cc to alsasymboltable_linux.cc
from alsasymboltable.h to alsasymboltable_linux.h
from audio_device_linux_alsa.cc to audio_device_alsa_linux.cc
from audio_device_linux_alsa.h to audio_device_alsa_linux.h
...etc

Modified files:
audio_device.gyp and audio_device_impl.cc : 
adapt to the new file names.

Test=none
Bug=none
Review URL: http://webrtc-codereview.appspot.com/91017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 12:41:56 +00:00
kma@google.com
8d0c98b954 some optimization in aecm for armv7. 
Review URL: http://webrtc-codereview.appspot.com/89018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 23:31:24 +00:00
kma@google.com
1489785326 Commit as part of http://webrtc-codereview.appspot.com/89018/ 
Committing separately as part of an OWNERS test.


git-svn-id: http://webrtc.googlecode.com/svn/trunk@345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 23:30:11 +00:00
xians@google.com
c3976c8c92 exclude the video_capture_module_test target from building with chromium. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=34
TEST=None
Review URL: http://webrtc-codereview.appspot.com/93015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 08:02:41 +00:00
andrew@webrtc.org
8910f278c5 Switch to webrtc.org accounts (for those which exist). 
Review URL: http://webrtc-codereview.appspot.com/97010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 05:16:31 +00:00
mikhal@google.com
9e9f86743c VP8 wrapper: Adding check for no show frames (NULL images) 
Review URL: http://webrtc-codereview.appspot.com/102001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 22:12:19 +00:00
wu@webrtc.org
1a023c95d8 Allow the NumberOfCodecs and GetCodec be called before the engine Init. 
VoE has the similar behavior, see VoECodecImpl::NumOfCodecs and VoECodecImpl::GetCode.
Review URL: http://webrtc-codereview.appspot.com/101005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 18:58:37 +00:00
andrew@webrtc.org
c68434d6a3 Add Chrome's exclusion filter to the Chrome build. 
These filters are now defined for our standalone build; if code is to rely on them, they must also be available in the Chrome build. (Chrome only defines them for non-third-party code).
Review URL: http://webrtc-codereview.appspot.com/101003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 18:14:47 +00:00
leozwang@google.com
22b699d2e0 Replace ANDROID with WEBRTC_ANDROID 
Review URL: http://webrtc-codereview.appspot.com/97008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 17:27:33 +00:00
mikhal@google.com
5ce0ff05ae Removing compiler warnings from video components 
Review URL: http://webrtc-codereview.appspot.com/101002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 17:17:39 +00:00
kma@google.com
579ee4d3f1 Neon optimization for an NS function. 
Review URL: http://webrtc-codereview.appspot.com/89017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 04:30:06 +00:00
ajm@google.com
8deee041b8 Enable full chromium_code warnings for all platforms. 
This is possible now that _linux files are used strictly on Linux.

Builds on Linux/Mac/Windows.
Review URL: http://webrtc-codereview.appspot.com/89010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 20:47:41 +00:00
ajm@google.com
4a4d7c6693 Fix a compile error on MSVC. 
- Declare all variables at the start of scope.
- Remove the needless local variables from these functions.
Review URL: http://webrtc-codereview.appspot.com/97009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 20:40:31 +00:00
mallinath@google.com
2cf5122efb VideoEngine core warnings remove. 
changes to VideoCaptureCapability is part of different CL.
Review URL: http://webrtc-codereview.appspot.com/98002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 17:57:54 +00:00
kma@google.com
881103225d optimization of resampling by 2 in ARMv7, in spl. 
Review URL: http://webrtc-codereview.appspot.com/92015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 15:40:10 +00:00
xians@google.com
f67f197049 exclude the neteq test targets from building with chromium. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=34
TEST=None
Review URL: http://webrtc-codereview.appspot.com/93017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 12:56:26 +00:00
xians@google.com
c3e7d30483 exclude vplib test target from building with chromium 
BUG=http://code.google.com/p/webrtc/issues/detail?id=34
Test=None
Review URL: http://webrtc-codereview.appspot.com/92010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 12:52:47 +00:00
xians@google.com
508eba5321 exclude vp8 and test_framework test targets from chromium. 
BUG=http://code.google.com/p/webrtc/issues/detail?id=34
Test=None
Review URL: http://webrtc-codereview.appspot.com/95012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 12:48:13 +00:00
xians@google.com
8706256eac exclude the test target from building with chromiurm. 
Review URL: http://webrtc-codereview.appspot.com/89016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 12:08:15 +00:00
xians@google.com
4257b175f3 The Cl is to support mixing output file in a stereo stream. 
Previously, an assert will be triggered in case it is not a mono stream.
With the CL, the mono file stream will be copied into a strereo stream and mixed with the channel stream.

More detail about the fix please refer to 
http://code.google.com/p/webrtc/issues/detail?id=36
Review URL: http://webrtc-codereview.appspot.com/93020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 12:02:36 +00:00
xians@google.com
0922815173 exclude the test target from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/93016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 11:44:07 +00:00
xians@google.com
0b0665acc1 This CL changes all the freq relevant variables to be int type. So it will take away the VoE "comparison between signed and unsigned integer expressions" warnings. 
BR,
/SX
Review URL: http://webrtc-codereview.appspot.com/89014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-08 08:18:44 +00:00
leozwang@google.com
81520b7955 Enable protobuf for apm module 
Review URL: http://webrtc-codereview.appspot.com/93019

git-svn-id: http://webrtc.googlecode.com/svn/trunk@319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 22:20:03 +00:00
marpan@google.com
8726e270ec Build error fix from r316/r314. 
Review URL: http://webrtc-codereview.appspot.com/95013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 21:38:56 +00:00
marpan@google.com
7bb8a9444d Change to r315: code to return error for 
incomplete frames (for _feedbackMode off) 
is commented out.
Review URL: http://webrtc-codereview.appspot.com/91016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 21:20:29 +00:00
leozwang@google.com
79835d1bd3 Clean up Android.mk 
Review URL: http://webrtc-codereview.appspot.com/92014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 21:01:02 +00:00
marpan@google.com
a1ec48dec2 Updates to VP8 wrapper: 
1) Added VP8E_SET_STATIC_THRESHOLD which sets threshold for
   forcing static macroblocks to skipped mode
   (useful for encoding of static background).
2) Removed some code in decoding to allow for 
   decoding of incomplete frames.
3) Minor change to MaxIntraTarget()
4) Added control setting for maximum (target) size of 
   intra-frame.
5) Increased kf_max_dist(reduce/remove key-frame periodicity)  
   since we use kf_auto_mode. 
Review URL: http://webrtc-codereview.appspot.com/101001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 20:51:32 +00:00
ajm@google.com
9f726b9d78 Resubmit r300 with changes required for Windows build. 
Review URL: http://webrtc-codereview.appspot.com/97005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 16:46:33 +00:00
xians@google.com
88f94f985e exclude the ACM test target from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/95011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 16:14:51 +00:00
xians@google.com
4d79bf164b exclude the g722 test target from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/92011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 16:14:25 +00:00
xians@google.com
e1cfd5f252 exclude test target from chromium 
Review URL: http://webrtc-codereview.appspot.com/92012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 16:14:02 +00:00
xians@google.com
d9102ca593 exclude the jpeg test target from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/92009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 15:45:52 +00:00
xians@google.com
7fa20c3880 exclude both audio_device_test_api and audio_device_test_func test targets from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/97006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 12:04:30 +00:00
xians@google.com
3d292f1649 this CL will strip the test targets from building with chromium. 
Review URL: http://webrtc-codereview.appspot.com/99004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 08:32:13 +00:00
xians@google.com
15ad55b03c This CL fixes the gclient runhooks problem with building with chromium. 
The problem is 
src/third_party/webrtc/third_party/protobuf/protobuf.gyp not found
Review URL: http://webrtc-codereview.appspot.com/92008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@306 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 08:21:25 +00:00
ronghuawu@google.com
102944a22b Revert r300. 
Review URL: http://webrtc-codereview.appspot.com/91014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 20:48:36 +00:00
xians@google.com
88bd440ef6 Removing the "initialized after" warnings. 
This CL tweat the order of the initialization in the constructor to
adapt to the order of declaration of the members.
Review URL: http://webrtc-codereview.appspot.com/92007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 15:33:30 +00:00
bjornv@google.com
55ce2d8a25 This changelist is basically a code cleanup and restructuring. Main things is that we now have a function for time to frequency transformation and the delay estimator is separated into its own struct and files. 
Review URL: http://webrtc-codereview.appspot.com/90004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 07:16:54 +00:00
ajm@google.com
ce7c2a231e Add SSE2 support for Windows. 
The previous defines controlling SSE2 use assumed GCC. To fix this:
- Import a chunk of defines to typedefs.h from Chrome's build_config.h, primarily to get WEBRTC_ARCH_X86_FAMILY.
- Add a check derived from WebP to define WEBRTC_USE_SSE2.
- Modify cpu_features.cc to work with MSVC. This code is borrowed from chrome/src/base/cpu.cc.
- Change AEC defines to use WEBRTC_USE_SSE2.
- Remove disable_sse2 check from aec.gyp. This is handled by WEBRTC_USE_SSE2.

(Also remove a bit of unused code from aec_core.h)
Review URL: http://webrtc-codereview.appspot.com/95008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 01:50:00 +00:00
leozwang@google.com
ce9bfbb33d Enable protobuf for Android 
Review URL: http://webrtc-codereview.appspot.com/91012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 23:34:31 +00:00
leozwang@google.com
094434fd33 Replace ANDROID with WEBRTC_ANDROID [video capture] 
Review URL: http://webrtc-codereview.appspot.com/99001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 21:37:33 +00:00
ajm@google.com
808e0e0dac Update the debug recordings to use protobufs. 
Also modify the unittest proto based to correspond with the changes. process_test is a bit of a hack job, but it works fine and isn't too unreadable. We should refactor it properly later.
Review URL: http://webrtc-codereview.appspot.com/98007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 21:08:51 +00:00
mikhal@google.com
320813c2d5 media_opt: Adding UEP to the hybrid mode 
Review URL: http://webrtc-codereview.appspot.com/89013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 20:47:50 +00:00
xians@google.com
22963abffe Removing the "initialized after" warnings. 
This CL tweat the order of the initialization in the constructor to adapt to the order of declaration of the members.
Review URL: http://webrtc-codereview.appspot.com/99002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 12:40:23 +00:00
holmer@google.com
e0f7d7b7e8 Fixing warnings in the VCM tests. Also fixes a couple of bugs related 
to those warnings.
Review URL: http://webrtc-codereview.appspot.com/98003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-03 07:49:56 +00:00
mallinath@google.com
af874ebc90 warning cleanup in render module 
Review URL: http://webrtc-codereview.appspot.com/95007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 22:52:33 +00:00
mallinath@google.com
df11c11cf2 Review URL: http://webrtc-codereview.appspot.com/97002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 22:42:23 +00:00
leozwang@google.com
56f4eb2b28 Replace ANDROID with WEBRTC_ANDROID 
Review URL: http://webrtc-codereview.appspot.com/89012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 22:24:43 +00:00
leozwang@google.com
d4e72f4ceb Add return value 
Review URL: http://webrtc-codereview.appspot.com/98004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 22:13:36 +00:00
marpan@google.com
c4df42be6b Added option for enabling the video protection settings and testing packet loss/delay in auto_test_loopback. 
Also allowing for setting frame size and start rate.
Review URL: http://webrtc-codereview.appspot.com/91008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-02 18:53:33 +00:00
frkoenig@google.com
e91655f8e2 Incorrect parameters being passed to trace function. 
And a small spelling error.
Review URL: http://webrtc-codereview.appspot.com/98001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 23:14:32 +00:00
mikhal@google.com
679450f4a6 media_opt_util: Update robustness settings for Hybrid mode. Updated table for the computation of the adjustment factor. 
Review URL: http://webrtc-codereview.appspot.com/93013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 22:14:58 +00:00
marpan@google.com
5fc2dcd64a Change to make the VP8-RTP Fragmentation (FI bits) setting (in the payload header) 
agree with "draft-westin-payload-vp8-02" document.

This issue was raised in: http://code.google.com/p/webrtc/issues/detail?id=31 
Review URL: http://webrtc-codereview.appspot.com/92005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 21:47:46 +00:00
marpan@google.com
191b780741 Added a correction factor to FEC overhead in media_opt_util. 
This is too handle cases of rate-mismatch (at low rates/low packet number) between estimate in mediaOpt and actual FEC generated in RTP.
Review URL: http://webrtc-codereview.appspot.com/93012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 19:59:57 +00:00
leozwang@google.com
f1ed5ad038 Replace ANDROID with WEBRTC_ANDROID 
Review URL: http://webrtc-codereview.appspot.com/91010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 17:36:09 +00:00
ajm@google.com
b5c49ff8af Rename all files/classes shared by Linux and Mac to Posix. 
Besides being a good idea in general, this lets us use the good chromium_code==1 warning settings on Mac and Windows. Straight rename except for a change to use os_posix in the gyp files.
Review URL: http://webrtc-codereview.appspot.com/88011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 17:04:04 +00:00
mikhal@google.com
b29d940db7 VCM: Updating Media Opt: 
1. Removed protection method specific code from SetTargetRates
2. Updated encoding rate following protection settings
3. Removing RTT max threshold from NACK, as it is not used in the receiver side.
4. Two bug fixes: FEC conversion function fix (line #133) and residual loss calculation (line #94) 
5. Removing compiler warnings
6.. Removed unused code and general clean-up. 
Review URL: http://webrtc-codereview.appspot.com/96002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 16:39:20 +00:00
ajm@google.com
02f8bbdbb7 Update the Android Makefile to use the new Dummy file. 
Review URL: http://webrtc-codereview.appspot.com/93011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 16:13:23 +00:00
ajm@google.com
7d962ccd32 Fix an error which causes an acccess violation. 
For 16-bit files, this was looping twice as long as necessary and reading outside the buffer.
http://code.google.com/p/webrtc/issues/detail?id=33
Review URL: http://webrtc-codereview.appspot.com/93007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-01 16:12:43 +00:00
ajm@google.com
81b9e2d2cc Revert an accidental change to prefer_fixed_point. 
Skipping pre-submit checks.
Review URL: http://webrtc-codereview.appspot.com/89009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-30 01:33:22 +00:00
frkoenig@google.com
dcfa09ac61 Correct parameters in WEBRTC_TRACE functions. 
Incorrect number of parameters resulted in complaints about
using unitialized memory when outputing with sprintf.
Review URL: http://webrtc-codereview.appspot.com/89008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 22:55:52 +00:00
vikasmarwaha@google.com
210d9fa08b incorrect return value after bind failure 
Review URL: http://webrtc-codereview.appspot.com/97001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 22:29:15 +00:00
mikhal@google.com
dbc9d5c5f1 video_processing: Removing compiler Warnings 
Review URL: http://webrtc-codereview.appspot.com/91009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 20:23:29 +00:00
mikhal@google.com
506bc3cf12 video_coding: Removing compiler warnings 
Review URL: http://webrtc-codereview.appspot.com/88010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 20:23:18 +00:00
mikhal@google.com
7acd3ba9e9 common_video: Removing compiler warnings 
Review URL: http://webrtc-codereview.appspot.com/93009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 20:23:01 +00:00
ajm@google.com
e89f6b5574 Create an empty device utility dummy, in order to stop using _linux suffixed files on non-Linux OSes. 
Besides being a good idea in general, this lets us use the good chromium_code==1 warning settings on Mac and Windows. (chromium_code automatically excludes _linux suffixed files on non-Linux.)
Review URL: http://webrtc-codereview.appspot.com/93010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 18:03:57 +00:00
ajm@google.com
f8dc8dc5f6 Generate protobuf classes at build-time. 
This method is well-established in Chromium. The new code is largely boilerplate copied from there. The advantage is that we don't have to maintain various versions of the classes; we just generate against whatever compiler version happens to exist at build-time.
Review URL: http://webrtc-codereview.appspot.com/93008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-29 17:29:08 +00:00
leozwang@google.com
f2c1a7b85c Clean up ANDROID macro definitions [audio_coding] 
Review URL: http://webrtc-codereview.appspot.com/91006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 20:39:08 +00:00
ajm@google.com
b0d9f3e6a3 Fix an ambiguous call to pow() error. 
Switch to powf() and explicitly define the second parameter as float.
Review URL: http://webrtc-codereview.appspot.com/89006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 18:43:18 +00:00
ajm@google.com
af61f6d4d4 Enable chromium_code==1 in a standalone build to have all the good warning settings from Chromium's common.gypi. Our third party code will still have the reduced warning settings. 
Review URL: http://webrtc-codereview.appspot.com/89001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 18:23:31 +00:00
marpan@google.com
11d986a68c Update to media_opt_util: 
(1) update to off-line table for fec 
(2) corresponding update and some code-cleanup for  
    FecProtectionFactor()
Review URL: http://webrtc-codereview.appspot.com/93006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 17:42:57 +00:00
ajm@google.com
59e41405d1 Add a fixed-point profile to the APM unit test. 
It uses fixed-point NS, AECM and adaptive digital AGC. It's selected by enabling "prefer_fixed_point" in common.gypi.
Review URL: http://webrtc-codereview.appspot.com/88009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 17:34:04 +00:00
ajm@google.com
11791b23f7 Fix some compiler warnings in fixed-point NS. 
Review URL: http://webrtc-codereview.appspot.com/93004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 17:33:51 +00:00
marpan@google.com
0d6fd2a943 Allowing for setting the video protection methods in auto_test_custom_call. 
Review URL: http://webrtc-codereview.appspot.com/96001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 17:15:39 +00:00
ajm@google.com
bb93f1d001 Fix "converting to non-pointer type from NULL" warnings. 
Review URL: http://webrtc-codereview.appspot.com/93005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-27 23:58:56 +00:00
leozwang@google.com
77f49d8219 Clean up ANDROID macro definitions [common_audio] 
Review URL: http://webrtc-codereview.appspot.com/91004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-27 18:58:33 +00:00
leozwang@google.com
842b8c3fe2 Build protobuf from source on Android 
Review URL: http://webrtc-codereview.appspot.com/95005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 23:04:24 +00:00
leozwang@google.com
fa23ea123f Clean up ANDROID macro definitions [common_video] 
Review URL: http://webrtc-codereview.appspot.com/91005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 21:22:27 +00:00
ajm@google.com
aace5b6bac Fix some apmtest errors, and add a path parameter. 
Review URL: http://webrtc-codereview.appspot.com/88007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 21:02:24 +00:00
leozwang@google.com
b3b87d7795 Add license statement 
Review URL: http://webrtc-codereview.appspot.com/92004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 18:31:26 +00:00
leozwang@google.com
b352700ff7 Clean up ANDROID macro definitions [system_wrapper] 
Review URL: http://webrtc-codereview.appspot.com/94001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 17:29:38 +00:00
marpan@google.com
13955743b0 Code cleanup for residual packet loss function in media_opt_util.cc. 
Review URL: http://webrtc-codereview.appspot.com/89004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 16:47:11 +00:00
leozwang@google.com
694019b9aa Clean up ANDROID macro definitions [audio_conf] 
Review URL: http://webrtc-codereview.appspot.com/94002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 05:04:12 +00:00
leozwang@google.com
03bc96d69a Clean up ANDROID macro definitions [video_render] 
Review URL: http://webrtc-codereview.appspot.com/95004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-26 05:03:10 +00:00
cduvivier@google.com
0e07d82f47 Optimization of 'cft1st': 
* scalar optimization, vectorization.
* 1.7% AEC overall speedup for the straight C path.
* 12.2% AEC overall speedup for the SSE2 path.
Review URL: http://webrtc-codereview.appspot.com/88006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-25 23:54:20 +00:00
ajm@google.com
26b83b581c Remove DTX from the ViE custom call test configuration. We don't normally recommend enabling this as it reduces audio quality. 
Review URL: http://webrtc-codereview.appspot.com/95003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-25 22:53:23 +00:00
leozwang@google.com
58d09cfb24 Clean up ANDROID macro definitions [audio_processing] 
Review URL: http://webrtc-codereview.appspot.com/95001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-25 22:49:00 +00:00
leozwang@google.com
3a2be19453 Enable unit_test for android 
Review URL: http://webrtc-codereview.appspot.com/95002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-25 22:47:56 +00:00
ajm@google.com
2098988d79 Tiny documentation fix. 
Review URL: http://webrtc-codereview.appspot.com/93003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-25 15:31:13 +00:00
mikhal@google.com
7740888ad6 VCM: Adding the fecMethod as a member of the hybrid NACK/FEC class. 
Review URL: http://webrtc-codereview.appspot.com/93002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 22:05:25 +00:00
mikhal@google.com
069e63085e Jpeg: Fixing encoder memory issue 
Review URL: http://webrtc-codereview.appspot.com/93001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 22:05:09 +00:00
ajm@google.com
da03e21908 Replace VAD with NS in comments. 
Review URL: http://webrtc-codereview.appspot.com/88005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 18:33:43 +00:00
frkoenig@google.com
e9a53a236b Fix writing outside of allocated space on bilinear frame scale. 
Was incorrectly extending filtered frame area by duplicating
the last pixel.  This was necessary when this was used with
color convert, but isn't needed for this application.

Added comments to help clarify some buffer usage.
Review URL: http://webrtc-codereview.appspot.com/84001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 17:24:11 +00:00
ajm@google.com
72610e22f8 Add output from an older protobuf compiler (version 2.3), to support the version in the Android tree. 
Review URL: http://webrtc-codereview.appspot.com/88004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 00:55:00 +00:00
wjia@google.com
e7d0c7cb53 fix gyp when build with Chromium. 
Review URL: http://webrtc-codereview.appspot.com/89003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-22 00:49:35 +00:00
leozwang@google.com
081b9f9628 Remove redundant spl header file 
Review URL: http://webrtc-codereview.appspot.com/92001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-21 21:28:29 +00:00
wjia@google.com
647aa16f0d exclude pulse audio when building with Chromium. 
BUG=none
TEST=local
Review URL: http://webrtc-codereview.appspot.com/91001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-21 20:45:31 +00:00
marpan@google.com
1b43b6d416 Changing the default VP8 packetization mode setting to kAggregate and balanced, from the previous settig of kStrict and balanced. 
The previous kStrict mode could generate very small packets when the encoded frame is smaller than MTU size. kAggregate will instead encapsulate whole frame into one packet if frame size is below MTU (and so will not generate too small packets), and otherwise it will separate out the first partition as in kStrict mode.

The balanced setting for kAggregate (from default of un-balanced) is also desirable, as equal size packets (for the first and remaining partition) should generally be more favorable for FEC.
Review URL: http://webrtc-codereview.appspot.com/89002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-21 16:49:54 +00:00
mikhal@google.com
022716b3af VCM/Media Opt Util: 1. Updating hybrid protection settings 2. Removing score 3. adding Fec as suffix to _residualPacketLoss for clarity. 4. Some clean up. 
Review URL: http://webrtc-codereview.appspot.com/88001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-20 23:12:57 +00:00