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43 Commits

Author SHA1 Message Date
perkj@webrtc.org
94cfde7c66 Removed scoped_refptr from libjingle.gyp 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 14:26:41 +00:00
perkj@webrtc.org
7e08613bda Move refcount and scoped_refptr to merge with libjingle. Deleted scoped_refptr_msg.h. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 14:26:25 +00:00
perkj@webrtc.org
aa32319046 Implement unittest for proxies of MediaStreamTrackInterface and MediaStreamInterface. 
This cl also change MediaStreamProxy to only allow setting the state from the signaling thread.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/237001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 09:32:38 +00:00
mallinath@webrtc.org
96ba19034c ref_count.h file name changed to refcount.h to keep as other ( most ) files are named in libjingle. 
Review URL: http://webrtc-codereview.appspot.com/240008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 08:01:11 +00:00
henrike@webrtc.org
0d55c8f96d Adding peerconnection_unittest. 
Review URL: http://webrtc-codereview.appspot.com/226004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 21:12:45 +00:00
mallinath@webrtc.org
5cb3064642 The change will separate the media tracks based on media type. MediaStreamInterface currently will have list for audio and video. This way we don't need to check for the track type before converting to respective mediatrack. 
Review URL: http://webrtc-codereview.appspot.com/230003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 13:19:08 +00:00
perkj@webrtc.org
63257d4bd2 Implement proxy for both audio and video tracks. 
The purpose of the proxy is that all calls to MediaStreamTracks should be done on the signaling thread.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/225004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-17 11:39:09 +00:00
henrike@webrtc.org
03a86998cd Fixes for build errors introduced most likely earlier today. 
Review URL: http://webrtc-codereview.appspot.com/228003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 23:36:38 +00:00
wu@webrtc.org
0c378112ec Define NO_SOUND_SYSTEM for chromium build. 
Review URL: http://webrtc-codereview.appspot.com/226002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 22:35:01 +00:00
wu@webrtc.org
ebc405d9c6 Remove the fakeportallocator from the libjingle.gyp. 
Review URL: http://webrtc-codereview.appspot.com/228001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 18:36:04 +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
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
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
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
perkj@webrtc.org
666f56bd41 MediaStreamHandler implements eventhandlers for streams and tracks. 
Sets local and remote renderer and capture device.

BUG=
TEST=

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
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.

BUG=
TEST=

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
wu@webrtc.org
78083bf750 * Add Serialize functions to PeerConnectionMessage. 
* Separated file for PeerConnectionMessage.
* Update to the latest and fix compiling errors
Review URL: http://webrtc-codereview.appspot.com/182002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 19:11:52 +00:00
mallinath@webrtc.org
9a1249d9e0 first cut of webrtcsession. Doesn't do much other than creating files and empty function bodies. 
Review URL: http://webrtc-codereview.appspot.com/186002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 18:15:21 +00:00
perkj@webrtc.org
2f56ff48a4 Implementation of PcSignaling. A Class to handle signaling between peerconnections. 
Review URL: http://webrtc-codereview.appspot.com/149002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-27 20:35:37 +00:00
perkj@webrtc.org
679e64d1fc Cleaning up of Peerconnection API. 
Removing RemoteMediaStream. Adding one universal implementation of MediaStream that is used for both remote and local media streams.
Removed AudioDevice and VideoDevice since VideoCaptureModule and AudioDeviceModule now is reference counted.
Changes LocalAudioTrackImpl and LocalVideoTrackImpl to AudioTrackImpl and VideoTrackImpl so they can be used to repressent both remote and local tracks.
Renamed files to a better name.
Review URL: http://webrtc-codereview.appspot.com/151001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 08:21:22 +00:00
wu@webrtc.org
c49db5ea48 The files included in devicemanager.h/cc still have some conflict with chromium. Let's keep the devicemanager mods for now and I will see how can we solve this next. 
Review URL: http://webrtc-codereview.appspot.com/166001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-20 00:40:52 +00:00
wu@webrtc.org
cb99f78653 * Update to use libjingle r85. 
* Remove (most of) local libjingle mods. Only webrtcvideoengine and webrtcvoiceengine are left now, because the refcounted module has not yet been released to libjingle, so I can't submit the changes to libjingle at the moment.
* Update the peerconnection client sample app.
Review URL: http://webrtc-codereview.appspot.com/151004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@625 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-19 21:59:33 +00:00
wu@webrtc.org
b27f3f16b6 Update to use the new opensource jsoncpp and remove jsoncpp mods. 
Review URL: http://webrtc-codereview.appspot.com/145001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-14 23:26:00 +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
2d9af90116 Fix error when building Peerconnection in Chrome. 
The error is due to wrong include path.
Review URL: http://webrtc-codereview.appspot.com/139016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 08:35:36 +00:00
perkj@google.com
e5ea75254f New Peerconnection manager implementation. Ready for review. 
Review URL: http://webrtc-codereview.appspot.com/134004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 07:25:56 +00:00
wu@webrtc.org
5a15ab9e36 Move the WebRtcDeviceManager and WebRtcMediaEngine to libjingle. 
Review URL: http://webrtc-codereview.appspot.com/139009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 23:04:52 +00:00
tommi@webrtc.org
87c546e89b Remove peerconnectionimpl_callbacks.h from libjingle.gyp. 
This file has actually never existed in trunk, but the 
line in libjingle.gyp wasn't removed when we decided not
to check in the file.  (see http://webrtc-codereview.appspot.com/60008/)
Review URL: http://webrtc-codereview.appspot.com/139011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-01 15:55:15 +00:00
perkj@google.com
3fcabbe45c Modified include path after after moving files to webrtc_dev. 
Review URL: http://webrtc-codereview.appspot.com/137010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 07:44:18 +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
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
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
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
mallinath@webrtc.org
467b1a9e4a Review URL: http://webrtc-codereview.appspot.com/116007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-17 00:07:03 +00:00
ronghuawu@google.com
492dbc258e Use the full path instead of the current directory. 
In chromium build this libjingle.gyp will be included by third_party/libjingle/libjingle.gyp. In that case the "." will mean the third_party/libjingle/ instead of what we want - third_party_mods/libjingle.
Review URL: http://webrtc-codereview.appspot.com/100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 00:36:01 +00:00
ronghuawu@google.com
35f534529b * Point the webrtc libjingle dependency to third_party_mods. 
* For unchanged files, change the third_party_mods libjingle.gyp to point to the original version of libjingle.
Review URL: http://webrtc-codereview.appspot.com/89015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-05 22:08:29 +00:00
ronghuawu@google.com
e256187f8b * Push the //depotGoogle/chrome/third_party/libjingle/...@38654 to svn third_party_mods\libjingle. 
* Update the peerconnection sample client accordingly.
Review URL: http://webrtc-codereview.appspot.com/60008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 17:44:30 +00:00
niklase@google.com
c7f3804131 git-svn-id: http://webrtc.googlecode.com/svn/trunk@171 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 09:35:19 +00:00
niklase@google.com
b849792667 git-svn-id: http://webrtc.googlecode.com/svn/trunk@169 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 09:15:38 +00:00
ronghuawu@google.com
e6988b9de5 * Update the session layer to p4 37930 
* Update the peerconnection_client in sync with updates on the libjingle side.
Review URL: http://webrtc-codereview.appspot.com/29008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@34 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-01 18:50:40 +00:00
ronghuawu@google.com
e8c5948b52 Revert back this change and wait when Tommi is only to submit the corresponding peerconnection test changes at the same time. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@32 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-01 17:14:19 +00:00
ronghuawu@google.com
7208ddddea Session layer update from p4 (cl37930) 
Review URL: http://webrtc-codereview.appspot.com/29008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@30 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-01 17:00:36 +00:00
niklase@google.com
5c61233a88 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:41:01 +00:00