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Author SHA1 Message Date
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.

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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.

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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.

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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. 
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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. 
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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.

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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
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.

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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

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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
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).

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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. 
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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
perkj@webrtc.org
1b6ff7adbe Connecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. 
This cl connects PeerConnectionImpl with WebrtcSession and MediaStreamHandlers.

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Review URL: http://webrtc-codereview.appspot.com/190005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:50:04 +00:00
perkj@webrtc.org
666f56bd41 MediaStreamHandler implements eventhandlers for streams and tracks. 
Sets local and remote renderer and capture device.

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Review URL: http://webrtc-codereview.appspot.com/192002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:55:17 +00:00
wu@webrtc.org
236fcaa89a Interface changes after we have the Serialize and Deserialize. 
Review URL: http://webrtc-codereview.appspot.com/186004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:34:19 +00:00
wu@webrtc.org
ed6d555775 * Add the crypto serialize and deserialize. 
* Populate candidates test data.

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Review URL: http://webrtc-codereview.appspot.com/190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:13:29 +00:00
mallinath@webrtc.org
ee2c391c15 more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. 
Review URL: http://webrtc-codereview.appspot.com/183005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 20:33:06 +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
perkj@webrtc.org
99239d5a41 First compiling version of peerconnection_client_dev using the new Peerconnection API. 
Links but does not work since the new peerconnection is under development.
I would like to commit a version with as few changes as possible to the old peerconnection_client but using the new PeerConnection API.

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Review URL: http://webrtc-codereview.appspot.com/183003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 15:59:40 +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.

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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