12 Commits

Author SHA1 Message Date
glaznev@webrtc.org
b5e60b6ca7 Remove non necessary check from WebSocket send function.
Peer connection may generate answer and ICE candidates before
websocket client is registered. Remove check from sendAnswer()
and sendLocalIceCandidate() functions and allow websocket client
to accumulate messages and send them later once it will be
registered.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44379004

Cr-Commit-Position: refs/heads/master@{#8508}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8508 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 19:19:39 +00:00
glaznev@webrtc.org
bc40324d9c Merge fixes and changed for Android AppRTCDemo from internal repo.
- Rename AppRTCDemoActivity to CallActivity and move UI controls
to a fragment.
- Add option to enable/disable statistics.
- Move peer connection and video constraints from URL to peer
connection client.
- Variable renaming.

R=jiayl@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33299004

Cr-Commit-Position: refs/heads/master@{#8319}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8319 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 23:05:04 +00:00
glaznev@webrtc.org
82415e395f Update AppRTCDemo to use renamed GAE messages.
BUG=4221
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 22:22:50 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
glaznev@webrtc.org
80452d70cb Sync Android AppRTCDemo with internal repo.
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.

R=jiayl@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:34:06 +00:00
glaznev@webrtc.org
f6a9714760 Remove peer connection and signaling calls from UI thread.
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.

R=jiayl@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 22:24:09 +00:00
wzh@webrtc.org
433006a6c2 Fixed style issues from lint and got rid of unused fields.
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7995 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 17:39:43 +00:00
glaznev@webrtc.org
b2bda67497 Removing old channel code from a few more places.
Plus adding peer connection close event.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7982 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:15:43 +00:00
glaznev@webrtc.org
e2a9261f3e Improve AppRTCDemo connection speed by sending all
http POST requests asynchronously.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 20:11:06 +00:00
glaznev@webrtc.org
369746bcb8 Support new WebSocket signaling format.
- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.

BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:28:52 +00:00
glaznev@webrtc.org
edc6e57a92 Support loopback mode and command line execution
for Android AppRTCDemo when using WebSocket signaling.

- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.

BUG=3995,3937
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 21:16:12 +00:00
henrik.lundin@webrtc.org
6f6ef72950 Add DCHECK to ensure that NetEq's packet buffer is not empty
This DCHECK ensures that one packet was inserted after the buffer was
flushed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-19 13:02:24 +00:00