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589 Commits

Author SHA1 Message Date
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=
TEST=

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=
TEST=

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=
TEST=

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=
TEST=

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