mikhal@webrtc.org
d24a97fae1
video coding test: deleting unused file(resampler_test.cc)
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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
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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:
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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.
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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.
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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.
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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.
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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
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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
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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
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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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
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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.
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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.
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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
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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.
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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.
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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.
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* 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
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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.
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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.
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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.
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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.
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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)
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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.
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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 []
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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.
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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.
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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.
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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.
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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.
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* 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.
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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).
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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.
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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
hellner@google.com
a386fc0a8b
Fixes build warnings due to unused variables.
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Code directly from http://code.google.com/p/webrtc/issues/detail?id=58 .
Review URL: http://webrtc-codereview.appspot.com/119007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@428 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 21:26:09 +00:00
hellner@google.com
9aa9996a19
Different solution than the one suggested in http://code.google.com/p/webrtc/issues/detail?id=56 however, should solve the same problem.
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Review URL: http://webrtc-codereview.appspot.com/126003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@427 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 21:25:55 +00:00
wu@webrtc.org
87c9b74b11
* Use the current thread as the signaling thread and worker thread to keep the unit test simple and easier to debug.
...
* I also merged the issue 113007.
This will be uploaded to the libjingle patch, so you may comment there if you want.
There's failure in the tests now, but I will let you review the threading change at the same time I will try to resolve the failure.
Review URL: http://webrtc-codereview.appspot.com/120002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@426 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 20:57:29 +00:00
wu@webrtc.org
ae53bf87d7
The variable ‘dummy’ set but not used being treated as errors in Fedora.
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Review URL: http://webrtc-codereview.appspot.com/122004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@425 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 18:56:56 +00:00
andrew@webrtc.org
ceb148ce59
Fix compile warnings in Release configuration.
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Review URL: http://webrtc-codereview.appspot.com/119003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@424 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 17:53:54 +00:00
mikhal@webrtc.org
06ad81fd58
video_coding: changing the UpdateMethod function (protection settings).
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Review URL: http://webrtc-codereview.appspot.com/126002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@423 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 16:03:59 +00:00
perkj@google.com
12f1fc4fe5
Fix initialization defect in constructor webrtc::ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(WebRtc_Word32, bool) initialization list.
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Review URL: http://webrtc-codereview.appspot.com/125002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@422 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 14:26:33 +00:00
perkj@google.com
a3fc1aac52
Fix Issue 59. Fix a constructor cast warning in video_X11_channel.cc.
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http://code.google.com/p/webrtc/issues/detail?id=59 .
Review URL: http://webrtc-codereview.appspot.com/122002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@421 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 13:38:14 +00:00
pwestin@webrtc.org
a070adbab2
Moved member RTPSender from private to protected.
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Review URL: http://webrtc-codereview.appspot.com/119006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@420 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-23 11:17:03 +00:00