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77 Commits

Author SHA1 Message Date
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
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
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.

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
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.

BUG=
TEST=

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. 
BUG=
TEST=

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.

BUG=
TEST=

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. 
BUG=
TEST=

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. 
BUG=
TEST=

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.

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
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

BUG=
TEST=

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.

BUG=
TEST=

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.

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
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.

BUG=
TEST=

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.

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
c93e36346b * Add Deserize for PeerConnectionMessage 
BUG=
TEST=

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();

BUG=
TEST=

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.

BUG=
TEST=

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
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
perkj@webrtc.org
0cc68dc38a Change Video capture module to be reference counting. Also prevent the module from beeing deleted using the interface. 
Furthermore remove all static module creation and deletion functions.
Review URL: http://webrtc-codereview.appspot.com/133012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-12 08:53:36 +00:00
mallinath@webrtc.org
c273019768 linking error after tommi's changes. 
Review URL: http://webrtc-codereview.appspot.com/140008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 19:34:15 +00:00
tommi@webrtc.org
73f98aebc6 Temporarily switch the numeric locale formatting to 'classic' while we process the signaling message. 
This is to avoid running into problems with jsoncpp and parts of libjingle
where we use STL and CRT routines that refer to the global locale for formatting.
If the current locale is e.g. Spanish, then numbers will be formatted as "12,34"
and not "12.34" as some parts (not all) of jsoncpp expect.

Code I noticed where we might run into this is here (but it's likely that there are
more places):

third_party\libjingle\source\talk\p2p\base\candidate.h (preference_str)
third_party_mods\libjingle\source\talk\app\webrtc\webrtc_json.cc
third_party\jsoncpp\src\lib_json\*writer*
third_party\jsoncpp\src\lib_json\*reader*

BUG=69
TEST=Verify that the signaling messages always contain numbers formatted as "1.23" and never "1,23" even though the regional settings on the machine specify otherwise (e.g. try setting it to Spanish).
Review URL: http://webrtc-codereview.appspot.com/140007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-08 17:46:46 +00:00
henrika@google.com
73d65513f1 Adds reference counting to the ADM. 
This CL modifies the ADM interface to ensure that an external ADM
can't call Create and Destroy any longer.

It also contains some minor style nits to conform better with
the Chromium style guide.
Review URL: http://webrtc-codereview.appspot.com/133014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@552 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-09-07 15:11:18 +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
wu@webrtc.org
b15bfd32d7 * Add the time_stamp as one parameter to the ViE ExternalRenderer interface. 
* Fix one issue in webrtcvideoengine where we should remove the renderer before adding a new one.
Review URL: http://webrtc-codereview.appspot.com/137011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-31 22:14:44 +00:00
mallinath@webrtc.org
f990eb3e88 Hi, 
Removed OnLocalStreamInitialized callback from the PeerConnection callback list. After adding OnAddStream trigger at the originator this callback was redundant. Also other modification is to provide same stream label in OnAddStream callback at the originator which provided in AddStream API.
Review URL: http://webrtc-codereview.appspot.com/138002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-30 17:16:35 +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
perkj@google.com
4094c49ddf Temporarily use digital AGC in WebRTC since Chromium can't support analog AGC. 
Fix suggested by henrika.
Review URL: http://webrtc-codereview.appspot.com/121001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 07:36:28 +00:00
mallinath@webrtc.org
92bace1faf Hi, 
This CL will support negotiation of RTCP Mux feature. Earlier we were by default enabling and assuming remote end point will support this feature as well. This will also remove the maintaining of transport channels in WebRtcSession. Its left to cricket::Transport
Review URL: http://webrtc-codereview.appspot.com/131005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-27 00:37:58 +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
hellner@google.com
b55c988b22 Updated peerconnection_unittest slightly. Also added it to the build. 
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