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31 Commits

Author SHA1 Message Date
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
5e160660a6 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). 
Second attempt to land https://webrtc-codereview.appspot.com/32399004/

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 20:35:13 +00:00
buildbot@webrtc.org
34bda43fa6 (Auto)update libjingle 79326895-> 79329222 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:44:55 +00:00
henrika@webrtc.org
e5421e9602 Volume buttons in AppRTCDemo should affect output audio volume. 
BUG=3279
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:19:19 +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
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc 
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 22:20:11 +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
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
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
glaznev@webrtc.org
58202946a7 Cleaning up Android AppRTCDemo. 
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.

BUG=
R=braveyao@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 17:42:38 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
buildbot@webrtc.org
21b4da8ebd (Auto)update libjingle 71753329-> 71766184 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 19:07:53 +00:00
glaznev@webrtc.org
c3288c130d Add OpenGL Android video renderer which can display multiple 
yuv420 images in a single GLSurfaceView.
Start using new video renderer in AppRTC demo app.

BUG=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 21:57:46 +00:00
tkchin@webrtc.org
56d114627b Fix AppRTC target configuration in libjingle_examples.gyp. 
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.

R=andrew@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/18489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:04:39 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo. 
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 22:26:06 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track. 
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 23:26:01 +00:00
kjellander@webrtc.org
190b72a350 Make libjingle Android example build without sourcing envsetup.sh 
See https://webrtc-codereview.appspot.com/11799004
for full details (separate to avoid webrtc+talk changes in same CL).

BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 08:35:49 +00:00
fischman@webrtc.org
7fa1fcb72c AppRTCDemo(ios): style/cleanup fixes following cr/62871616-p10 
BUG=2168
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/9709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:11:56 +00:00
henrike@webrtc.org
d3d6bce9ed (Auto)update libjingle 62865357-> 62871616 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 20:41:22 +00:00
henrike@webrtc.org
28da47c52f Android example apps: fixes issue where useful failure information was suppressed. 
BUG=2808
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 19:03:51 +00:00
henrike@webrtc.org
2ce9a64b75 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. 
BUG=12545067
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:49:53 +00:00
fischman@webrtc.org
000dde99c8 Android build: make it quiet on success and not overly noisy on failure. 
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6199005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 22:49:35 +00:00
fischman@webrtc.org
df7b1d6e39 AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure. 
Also silence a 'cd' that would otherwise emit the path/to/talk.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-11 22:36:22 +00:00
fischman@webrtc.org
1977960866 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 16:48:51 +00:00
fischman@webrtc.org
ddc5a19ce9 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. 
BUG=2458
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2348004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:09:40 +00:00
fischman@webrtc.org
e3de6b1e90 Enable ObjC build by default and reenable 64-bit mac libjingle build 
BUG=2124
TESTED=trybots & building for mac, mac64, ios-sim, and ios-device on my MBP all build everything in out/Debug.
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2080004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 19:31:21 +00:00
fischman@webrtc.org
dd14b2add1 libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. 
- $JAVA_HOME / java_home missing or not pointing to a JDK
- Multiple or zero mac codesigning identities

BUG=2206
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2012004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 18:06:29 +00:00
fischman@webrtc.org
1bc1954174 AppRTCDemo: builds using ninja on iOS for simulator and device! 
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
  running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
  the hand-crafted Xcode project (which has never worked in its checked-in
  form), including a gyp action to sign the sample app for deployment to an iOS
  device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
  (in a surprising twist of fate, the API landed quite a bit later than the
  sample app and this is the first time the CR-time changes in the API are
  reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
  the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
  from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
  ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
  formatting craxy.

BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 18:29:45 +00:00
fischman@webrtc.org
147d44a450 AppRTCDemo: replace the use of query-string parameters for pre-JB devices. 
Replaces the use of a query-string parameter with a (once-per-session)
JS-to-Java function call, because query-string parameters on file:// URLs are
busted on ICS and earlier Android releases
(https://code.google.com/p/android/issues/detail?id=17535).

Also added channel.html to the list of inputs to cause edits to it to cause a
rebuild of the .apk.

BUG=1949
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1890004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-29 19:07:33 +00:00
henrike@webrtc.org
723d683ecb Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1797004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 16:04:50 +00:00
henrike@webrtc.org
28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to 
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 00:45:36 +00:00