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Author SHA1 Message Date
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
perkj@google.com
4094c49ddf Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. 
Fix suggested by henrika.
Review URL: http://webrtc-codereview.appspot.com/121001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:36:28 +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
mallinath@webrtc.org
92bace1faf Hi, 
This CL will support negotiation of RTCP Mux feature. Earlier we were by default enabling and assuming remote end point will support this feature as well. This will also remove the maintaining of transport channels in WebRtcSession. Its left to cricket::Transport
Review URL: http://webrtc-codereview.appspot.com/131005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-27 00:37:58 +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
mallinath@webrtc.org
b62c776eca moving all new version related files to webrtc_dev and removed from webrtc. 
Review URL: http://webrtc-codereview.appspot.com/138001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 17:19:09 +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
tommi@webrtc.org
350d091e0e Send the hangup message when asked to disconnect from a peer. 
Review URL: http://webrtc-codereview.appspot.com/131006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-26 13:20:41 +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
b55c988b22 Updated peerconnection_unittest slightly. Also added it to the build. 
Review URL: http://webrtc-codereview.appspot.com/133003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 23:01:40 +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
hellner@google.com
b2801f3a16 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. 
Review URL: http://webrtc-codereview.appspot.com/131003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 21:37:08 +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
andrew@webrtc.org
aed0348e5b Roll gyp 985:1012 
Fix the world rebuilding in make 3.82.
http://code.google.com/p/webrtc/issues/detail?id=62

r1012 also allows Chrome to build with Make on Mac. Haven't tested WebRTC, but it would be nice to have.
Review URL: http://webrtc-codereview.appspot.com/119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 18:45:51 +00:00
hellner@google.com
40373cc184 Bugfix in unittest and some minor refactoring. 
Review URL: http://webrtc-codereview.appspot.com/137003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 17:17:30 +00:00
wu@webrtc.org
eb9572e501 Add the new peerconnection factory to the scons file. 
Review URL: http://webrtc-codereview.appspot.com/134001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 16:58:58 +00:00
niklas.enbom@webrtc.org
e129ae944e Review URL: http://webrtc-codereview.appspot.com/137002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 16:52:34 +00:00
hellner@google.com
3227ed567b Fixed potential memory leak in unit test and removed an unnecessary copy. 
Review URL: http://webrtc-codereview.appspot.com/131001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 15:34:19 +00:00
tommi@webrtc.org
102b2270c7 First version of the peerconnection client application for Linux. 
I made several updates to the Windows version as well so that both
implementations share
a big portion of the code.
The underlying PeerConnection notifications have changed a bit since the last
update
so that there's still a known issue that I plan to fix in my next change:

  // TODO(tommi): There's a problem now with terminating connections:
  // When ending a conversation, both peers now send a signaling message
  // that indicates that their ports are closed (port=0).  The trouble this
  // causes us here is that we can interpret such a message as an invite
  // to a new conversation.  So, currently there is a bug that ending
  // a conversation can immediately start a new one.
  // To fix this I plan to change how conversations start and have a special
  // notification message via the server that prepares a client for a
  // conversation instead of automatically recognizing the first signaling
  // message as an invite.
Review URL: http://webrtc-codereview.appspot.com/112008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 15:03:52 +00:00
tommi@webrtc.org
137ece4ac3 * Make GetReadyState accessible via the PeerConnection interface. 
* Update PeerConnection implementations to include "virtual"
in the method declarations.
* Add a check for a valid signaling thread in webrtcsession.cc.
Review URL: http://webrtc-codereview.appspot.com/137001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 14:18:25 +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
mallinath@webrtc.org
1cdc6b5d79 This CL adding a factory class which has the responsibility of creating peerconnection objects. This is very basic class doesn't do any reference count, user has the responsibility to delete the object externally. 
Review URL: http://webrtc-codereview.appspot.com/122006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 23:50:05 +00:00
hellner@google.com
d1015fe677 Replaced regular sleep with a talk_base::Thread::ProcessMessages(..) call so that Posts get some execution time from the main thread. 
Review URL: http://webrtc-codereview.appspot.com/122007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 21:35:09 +00:00
turajs@google.com
5cc9c68e8d Fixing a warning discovered while compiling with clang. 
Review URL: http://webrtc-codereview.appspot.com/120003

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 15:48:47 +00:00
perkj@google.com
accd686b31 Implementation of media streams. Work in progress. 
Review URL: http://webrtc-codereview.appspot.com/117002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 15:43:42 +00:00
stefan@webrtc.org
49cbc512ae Fix unused variable warning in video_coding. 
Issue 57: [Patch] Fix unused variable warnings in the video_coding module
Review URL: http://webrtc-codereview.appspot.com/126004

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 00:51:36 +00:00
wu@webrtc.org
9788e18532 * Add PeerConnectionProxy to forward all the API calls to signaling thread. 
* Use Send instead of Post so that we can report error.
Review URL: http://webrtc-codereview.appspot.com/113009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 23:49:44 +00:00
wjia@google.com
4482b04207 revert r430 to keep webrtc always ready to roll in chromium. 
r430 will be used when libvpx in chromium is rolled to Cayuga.
Review URL: http://webrtc-codereview.appspot.com/119008

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@430 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 23:08:30 +00:00
mallinath@webrtc.org
dec6aa57f3 This CL will remove sending any signal after calling Close and RemoveStream. I am thinking to remove Close method at all, since application can directly delete the object if it wants to end the call with all active streams. Will send that change later in a different CL. 
Review URL: http://webrtc-codereview.appspot.com/119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 22:17:03 +00:00