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23 Commits

Author SHA1 Message Date
Niklas Enbom
8a19f3dc62 Relanding https://webrtc-codereview.appspot.com/56589004 
BUG=
TBR=cpaulin@chromium.org

Review URL: https://codereview.webrtc.org/1176023002.

Cr-Commit-Position: refs/heads/master@{#9410}
 2015-06-10 18:39:34 +00:00
Niklas Enbom
54b0ca553f Revert "Landing https://webrtc-codereview.appspot.com/53669004/" 
This reverts commit 2aef19cbde01cb975eb3d6100610d31bbbae9258.

BUG=

TBR=cpaulin@chromium.org

Review URL: https://codereview.webrtc.org/1168313003.

Cr-Commit-Position: refs/heads/master@{#9404}
 2015-06-09 23:21:29 +00:00
Niklas Enbom
2aef19cbde Landing https://webrtc-codereview.appspot.com/53669004/ 
BUG=

Review URL: https://codereview.webrtc.org/1169123003.

Cr-Commit-Position: refs/heads/master@{#9403}
 2015-06-09 22:38:28 +00:00
glaznev@webrtc.org
e3fccd4268 Merge changes from internal repo to AppRTCDemo. 
- Add a setting option to disable outgoing video in a call.
- Add an option to select audio codec.
- Add an option to specify audio bitrate for Opus codec.
- Plus add an option to select H.264 as default video codec.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42449004

Cr-Commit-Position: refs/heads/master@{#8468}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-24 00:54:00 +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
44ae4c8b07 Support using VP9 video codec in AppRTCDemo. 
- Add peer connection Java API to initialize field trial string.
- Add setting option to select VP8 or Vp9 as default video codec.
- Minor code clean up and allowing 720p portrait encoding.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39899004

Cr-Commit-Position: refs/heads/master@{#8303}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 23:26:41 +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
jiayl@webrtc.org
5eb71eb4f4 Fix style issues from lint. 
BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-30 22:44:11 +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
jiayl@webrtc.org
a6f7ba6848 Add a AppRTCDemo setting to change the GAE server. 
BUG=4041
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 17:32:14 +00:00
jiayl@webrtc.org
16a05dddb8 Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. 
BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:12:03 +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
dea5173edf Add start bitrate and vp8 hw acceleration option to 
Android AppRTCDemo.

- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.

BUG=4046
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:02:13 +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
henrika@webrtc.org
2176db343c AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) 
This CL was incorrectly reverted in r7647 by the libjingle sync bot.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-18 13:22:28 +00:00
buildbot@webrtc.org
0ef890a4ba (Auto)update libjingle 79285346-> 79320771 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:22:08 +00:00
mcasas@webrtc.org
6340acde68 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. 
Also removed some unused "summary" ListPreference
fields.

The looks of it can be found in [1] (lowest row).

[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:05:48 +00:00
glaznev@webrtc.org
5f38c8d1b8 Android AppRTCDemo improvements: 
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.

BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 22:18:52 +00:00
glaznev@webrtc.org
243eb8e9af Adding setting screen to AppRTCDemo. 
- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.

BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 17:22:15 +00:00
perkj@webrtc.org
470988742a Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. 
BUG=3934
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 11:38:19 +00:00
glaznev@webrtc.org
7bb4a9881d Merging Henrik's and Peter's changes for AppRTCDemo 
from https://github.com/hkjellander/AppRTCDemo.

Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.

BUG=3938
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:43:37 +00:00