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105 Commits

Author SHA1 Message Date
perkj@webrtc.org
36a992b030 Merge streamparams and mediasession from libjingle and made necessary changes in peerconnection. 
-Removed ssrc from tracks.
-Updated PeerConnectionMessage parsing and serialization.

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Review URL: http://webrtc-codereview.appspot.com/239020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@856 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-11-01 11:46:56 +00:00
henrike@webrtc.org
8885d22399 Review URL: http://webrtc-codereview.appspot.com/239015 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-31 20:35:13 +00:00
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
35a12cdf60 Fix comment. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 15:09:03 +00:00
perkj@webrtc.org
8129752c3b Add refcount and scoped_refptr. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-28 15:08:54 +00:00
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
mallinath@webrtc.org
58b4f1c43c Fixes broken build in peerconnection unit tests 
Review URL: http://webrtc-codereview.appspot.com/244006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-24 05:10:55 +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.

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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
ca8b3a387e kind() method in track interface is changed to std::string to keep uniformity with other get methods 
Review URL: http://webrtc-codereview.appspot.com/242003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-21 09:18:25 +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
perkj@webrtc.org
2ebc9ce5a3 Fix broken PeerConnection Dev build. 
Fix MediaStreamHandler::CommitLocalStreams refactoring error.

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Review URL: http://webrtc-codereview.appspot.com/243005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-20 11:52:31 +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
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.

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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
c01c358f54 session/phone/channel.cc updates after new push of libjingle revision. 
Review URL: http://webrtc-codereview.appspot.com/225003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 09:45:24 +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
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
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
3a6d4f4268 Fix setting VideoCaptureModule and VideoRenderer for local and remote streams. 
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Review URL: http://webrtc-codereview.appspot.com/205002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-06 16:10:10 +00:00
mallinath@webrtc.org
fa41d807a8 Fixes session state transition and registering observer. 
Review URL: http://webrtc-codereview.appspot.com/203001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 22:49:59 +00:00
mallinath@webrtc.org
29787c71a0 Changes to WebRtcSession after Provider(s) interface addition. 
Review URL: http://webrtc-codereview.appspot.com/201001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-05 18:52:26 +00:00
perkj@webrtc.org
487e401a27 Moving creation of sessiondescriptions to webrtcsession. 
Fixing defect durin close down in peerconnectionmanager.

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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
perkj@webrtc.org
cb4ab65dfc Moved creation of objects to the signaling thread. 
Fixed defect of not initializing remote_media_streams in peerconnection_impl.cc
Fixed defect in glare case of peerconnectionsignaling.cc

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Review URL: http://webrtc-codereview.appspot.com/196001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-04 17:54:34 +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
1b6ff7adbe Connecting PeerConnectionImpl with WebrtcSession and MediaStreamHandlers. 
This cl connects PeerConnectionImpl with WebrtcSession and MediaStreamHandlers.

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Review URL: http://webrtc-codereview.appspot.com/190005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 22:50:04 +00:00
perkj@webrtc.org
666f56bd41 MediaStreamHandler implements eventhandlers for streams and tracks. 
Sets local and remote renderer and capture device.

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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
wu@webrtc.org
236fcaa89a Interface changes after we have the Serialize and Deserialize. 
Review URL: http://webrtc-codereview.appspot.com/186004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:34:19 +00:00
wu@webrtc.org
ed6d555775 * Add the crypto serialize and deserialize. 
* Populate candidates test data.

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Review URL: http://webrtc-codereview.appspot.com/190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 21:13:29 +00:00
mallinath@webrtc.org
ee2c391c15 more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state. 
Review URL: http://webrtc-codereview.appspot.com/183005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 20:33:06 +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
wu@webrtc.org
c93e36346b * Add Deserize for PeerConnectionMessage 
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Review URL: http://webrtc-codereview.appspot.com/189001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-30 18:08:51 +00:00
perkj@webrtc.org
e804ee1a80 This patch hooks up PeerConnectionImpl to PeerConnectionSignaling. 
Implements
virtual bool ProcessSignalingMessage(const std::string& msg);
virtual scoped_refptr<StreamCollection> remote_streams();
virtual void CommitStreamChanges();

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Review URL: http://webrtc-codereview.appspot.com/187001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-29 22:27:54 +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
5045f671d0 Add SignalUpdateSessionDescription to PeerConnectionSignaling. 
This is to allow webrtcsession to setup the mediachannels based on tracks.

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Review URL: http://webrtc-codereview.appspot.com/184001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-28 23:06:46 +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
ronghuawu@google.com
c389aa2615 Fix the bad video issue on Window client by increasing the rtp recv buffer size. 
Send buffer doesn't really matter, just to keep the same as talk does.

The same fix is submitted to libjingle for reivew. But I think it's worth to fix it here too as
it may take while for webrtc to get from libjingle. This patch is slightly different then that
one as I don't want to add the webrtcvideoengine.h back to webrtc.
Review URL: http://webrtc-codereview.appspot.com/166002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-21 16:53:45 +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