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Author SHA1 Message Date
henrike@webrtc.org
4d8c81878e The implementation before this change list keeps the ownership of memory that is used by peer connection instances in the peer connection manager. This means that if the peer connection manager is deleted before all peer connections it has created, these peer connections will be pointing to invalid memory. 
The solution in this CL is to create a bundle of the memory that needs to be alive as long as there are any peer connections or peer connection manager instances. This bundle is scoped reference counted so that it is deleted only when there are no references to it. This enables the peer connection and manager to be deleted in any order.
Review URL: http://webrtc-codereview.appspot.com/246003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 18:00:10 +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
mallinath@webrtc.org
f553ec70c7 Notifier and RefCount interface and implementation class name changed according to the naming convention. 
Review URL: http://webrtc-codereview.appspot.com/241003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 06:24:24 +00:00
perkj@webrtc.org
1305a1d05e Fix rendering in new PeerConnection API. 
Fix MediaStreamHandler to make sure it releases the reference to a renderer when it is no longer needed.
Removes the use of the signaling thread in MediaStreamHandler.

Fix renderering in peerconnection_client_dev. It now uses the reference counted render wrapper.

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Review URL: http://webrtc-codereview.appspot.com/242001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-18 11:54:46 +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.

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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
mallinath@webrtc.org
ebc0a00197 One of Justin comment was to have XXXXInterface and XXXX, rather than XXXX and XXXXImpl. So here are the changes, i don't like to call some the classes as interfaces like MediaStreamTrackListInterface, but they fit the criteria to be called as interface. 
Review URL: http://webrtc-codereview.appspot.com/226001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 07:04:02 +00:00
mallinath@webrtc.org
103f33b734 Changes after comments received from Justin and Harald. Few comments are not implemented like moving track implementation to base<> and then have child classes based on the type of track. 
Review URL: http://webrtc-codereview.appspot.com/217004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-13 14:31:20 +00:00
perkj@webrtc.org
6a34d584b8 Implement MediaStreamProxy. 
This implements a proxy for MediaStreams and MediaStreamTracklists.

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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
wu@webrtc.org
77d7d5455e Replace the DestroyDeviceInfo with a virtual destructor. 
Review URL: http://webrtc-codereview.appspot.com/212005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-12 16:57:53 +00:00
perkj@webrtc.org
38e400a967 Adding native client test page to test loopback. 
The test page is the same as the previouse test page but exchange offer messagesto answer messages.

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Review URL: http://webrtc-codereview.appspot.com/193005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-12 12:59:13 +00:00
wu@webrtc.org
ea89922b56 Add VideoCaptureFactory so that we don't need to expose VideoCaptureImpl. 
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Review URL: http://webrtc-codereview.appspot.com/213002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-11 17:13:51 +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
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
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