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107 Commits

Author SHA1 Message Date
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
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
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@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
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
hellner@google.com
b2801f3a16 Added the remaining test cases for the webrtcsession unittest also some minor refactoring. 
Review URL: http://webrtc-codereview.appspot.com/131003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 21:37:08 +00:00
hellner@google.com
40373cc184 Bugfix in unittest and some minor refactoring. 
Review URL: http://webrtc-codereview.appspot.com/137003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 17:17:30 +00:00
wu@webrtc.org
eb9572e501 Add the new peerconnection factory to the scons file. 
Review URL: http://webrtc-codereview.appspot.com/134001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 16:58:58 +00:00
hellner@google.com
3227ed567b Fixed potential memory leak in unit test and removed an unnecessary copy. 
Review URL: http://webrtc-codereview.appspot.com/131001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 15:34:19 +00:00
tommi@webrtc.org
137ece4ac3 * Make GetReadyState accessible via the PeerConnection interface. 
* Update PeerConnection implementations to include "virtual"
in the method declarations.
* Add a check for a valid signaling thread in webrtcsession.cc.
Review URL: http://webrtc-codereview.appspot.com/137001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-25 14:18:25 +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
hellner@google.com
d1015fe677 Replaced regular sleep with a talk_base::Thread::ProcessMessages(..) call so that Posts get some execution time from the main thread. 
Review URL: http://webrtc-codereview.appspot.com/122007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-24 21:35:09 +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
dec6aa57f3 This CL will remove sending any signal after calling Close and RemoveStream. I am thinking to remove Close method at all, since application can directly delete the object if it wants to end the call with all active streams. Will send that change later in a different CL. 
Review URL: http://webrtc-codereview.appspot.com/119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 22:17:03 +00:00
wu@webrtc.org
87c9b74b11 * Use the current thread as the signaling thread and worker thread to keep the unit test simple and easier to debug. 
* I also merged the issue 113007.

This will be uploaded to the libjingle patch, so you may comment there if you want.

There's failure in the tests now, but I will let you review the threading change at the same time I will try to resolve the failure.
Review URL: http://webrtc-codereview.appspot.com/120002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-23 20:57:29 +00:00
mallinath@webrtc.org
6f555dcafe Review URL: http://webrtc-codereview.appspot.com/119002 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 18:33:51 +00:00
wu@webrtc.org
eb29a9789d * Remove the previous renderer before set a new one. 
* Allow to unregister a renderer by giving a NULL point.
Review URL: http://webrtc-codereview.appspot.com/123001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@412 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-22 15:58:03 +00:00
mallinath@webrtc.org
bca7fa09af Review URL: http://webrtc-codereview.appspot.com/118001 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 16:39:18 +00:00
mallinath@webrtc.org
310689e26d Review URL: http://webrtc-codereview.appspot.com/115007 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 01:13:31 +00:00
wu@webrtc.org
765c918677 Changes based on the review comments. 
* Rename WebRTCSession to WebRtcSession.
* Add comments to the signal.
Review URL: http://webrtc-codereview.appspot.com/114009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-19 00:14:23 +00:00
mallinath@webrtc.org
bfc63ae83f Review URL: http://webrtc-codereview.appspot.com/113008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-18 18:01:31 +00:00
ronghuawu@google.com
18cec47eab Bug fix for OnRemoveStream. 
Review URL: http://webrtc-codereview.appspot.com/116009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-18 03:05:41 +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
wu@webrtc.org
74a49a833e Two changes: 
1) Libjingle and webrtc use different jsoncpp, so we need to distinguish them.
2) Update the webrtcsession_unittest:
 * Use the Thread::SleepMs
 * Remove main, which has been defined in talk/base/unittest_main.
Review URL: http://webrtc-codereview.appspot.com/107003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-12 00:20:24 +00:00
mallinath@google.com
424e76a8c1 Review URL: http://webrtc-codereview.appspot.com/97012 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-11 20:44:56 +00:00
andrew@webrtc.org
8910f278c5 Switch to webrtc.org accounts (for those which exist). 
Review URL: http://webrtc-codereview.appspot.com/97010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-10 05:16:31 +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
a852de7bae The new PeerConnection Api (under development) from p4 libjingle/...@38654. 
Review URL: http://webrtc-codereview.appspot.com/91002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 21:09:31 +00:00
ronghuawu@google.com
907c355ca3 Add owner file for the third_party_mods\libjingle. 
Review URL: http://webrtc-codereview.appspot.com/95010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-04 21:06:20 +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
ajm@google.com
98c5001e40 Reorganize libvpx to mimic Chromium's layout. We now sync libvpx to a subdirectory of third_party/libvpx where we've submitted our few wrapper files. This avoids having to "svn export" to the directory and allows direct committing of files. 
Review URL: http://webrtc-codereview.appspot.com/64001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-09 21:16:30 +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
leozwang@google.com
0b0c28c495 add android makefile, some modification in vpx makefile to build encoder from c source for now 
Review URL: http://webrtc-codereview.appspot.com/29012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@50 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-07 17:24:39 +00:00
ajm@google.com
06313d5de9 Fixing some incorrect file names in gyp files reported by an external user. See the gyp warnings at the bottom of this page: http://pastebin.com/4sdp5ivs 
I'm not sure how he got the warnings; I couldn't figure out how to display them myself.
Review URL: http://webrtc-codereview.appspot.com/22022

git-svn-id: http://webrtc.googlecode.com/svn/trunk@44 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-06 16:59:21 +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