Commit Graph

918 Commits

Author SHA1 Message Date
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
wu@webrtc.org
6c2d7107ae * Update to use the new libjingle release.
* Stop using any local mods for the default build (non-dev).
Review URL: http://webrtc-codereview.appspot.com/224001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@737 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 16:58:50 +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
mallinath@webrtc.org
103f33b734 Changes after comments received from Justin and Harald. Few comments are not implemented like moving track implementation to base<> and then have child classes based on the type of track.
Review URL: http://webrtc-codereview.appspot.com/217004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@735 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 14:31:20 +00:00
kjellander@webrtc.org
7951e819af Simple utility method for finding the project root dir (to be used by tests loading resource files)
The code has no intent to be superportable in all possible scenarios, since it will only be used by our own test code.
I reviewed more sophisticated libraries for doing similar things but came to the conclusion that they introduced more dependencies than motivated for this single purpose.

The unit test has been tested successfully executed on Linux (cmd line and Eclipse), Mac (XCode) and Windows (VS2008).

Review URL: http://webrtc-codereview.appspot.com/223002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@734 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 12:24:41 +00:00
perkj@webrtc.org
6a34d584b8 Implement MediaStreamProxy.
This implements a proxy for MediaStreams and MediaStreamTracklists.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/217003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@733 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-13 08:48:43 +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
perkj@webrtc.org
38e400a967 Adding native client test page to test loopback.
The test page is the same as the previouse test page but exchange offer messagesto answer messages.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@730 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-12 12:59:13 +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
tommi@webrtc.org
ed081a99a9 Print info about the local and remote resolution in the Windows client.
Review URL: http://webrtc-codereview.appspot.com/212001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@721 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 12:58:21 +00:00
perkj@webrtc.org
73ba4160f6 Fix OnClose(socket, NO_ERROR) compile error on Linux.
Merge Peerconnection_client_dev with Peerconnection_client.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/215002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@720 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 11:15:35 +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
f7b36a47c0 Fix bug in the server where a wait request was incorrectly handled.
Change the assert macro on Windows to make it easier to debug.
Review URL: http://webrtc-codereview.appspot.com/212002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@718 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 09:51:52 +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
tommi@webrtc.org
5a695d6094 Fix bug in the client that caused signaling messages to be dropped.
Also fixing potential out-of-order delivery of signaling messages.
Review URL: http://webrtc-codereview.appspot.com/214005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@716 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-10 08:16:26 +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
tommi@webrtc.org
c8c4deb0bb Fix Windows build. %zu isn't supported in the crt implementation
we use there, so it just crashes.
Review URL: http://webrtc-codereview.appspot.com/213001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@712 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-09 18:32:17 +00:00
tommi@webrtc.org
5a945ecc28 A little upgrade to the HTML test page:
* Signaling messages are added to the log with a '+' / '-' sign to expand/collapse the message.  This makes the log easier to read and each message can be read separate from the others.
* Loopback enabled by default since that's the most common use case.
* Wrapped some lines at 80 for easier future diffing.
Review URL: http://webrtc-codereview.appspot.com/214001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@711 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-07 13:23:11 +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
perkj@webrtc.org
f6ab63c08a Update PeerConnection_client to open a video capture device.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/205001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@707 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 20:36:23 +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
2915f6fc44 Use proper printf size_t specifier to fix Linux 32-bit build.
http://code.google.com/p/webrtc/issues/detail?id=97
Review URL: http://webrtc-codereview.appspot.com/204001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@704 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 16:37:03 +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
perkj@webrtc.org
3a6d4f4268 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/205002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@701 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-06 16:10:10 +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
mallinath@webrtc.org
fa41d807a8 Fixes session state transition and registering observer.
Review URL: http://webrtc-codereview.appspot.com/203001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@697 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 22:49:59 +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
mallinath@webrtc.org
29787c71a0 Changes to WebRtcSession after Provider(s) interface addition.
Review URL: http://webrtc-codereview.appspot.com/201001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@695 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 18:52:26 +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
perkj@webrtc.org
487e401a27 Moving creation of sessiondescriptions to webrtcsession.
Fixing defect durin close down in peerconnectionmanager.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/193004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@693 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-05 17:15:36 +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
perkj@webrtc.org
cb4ab65dfc Moved creation of objects to the signaling thread.
Fixed defect of not initializing remote_media_streams in peerconnection_impl.cc
Fixed defect in glare case of peerconnectionsignaling.cc

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/196001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@690 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 17:54:34 +00:00
mallinath@webrtc.org
bafca109db Temp hook in WebRtcSession to VideoChannel.
Review URL: http://webrtc-codereview.appspot.com/195001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@689 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-04 17:45:21 +00:00