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.
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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.
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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.
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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.
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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.
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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.
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* 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.
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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.
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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.
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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.
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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.
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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.
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* 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.
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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.
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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.
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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.
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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.
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* 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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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
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
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
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
mallinath@webrtc.org
467b1a9e4a
Review URL: http://webrtc-codereview.appspot.com/116007
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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.
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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.
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* 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.
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* 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
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* 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.
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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)
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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