Summary:
- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.
TBR=glaznev
BUG=4103,4109
Review URL: https://webrtc-codereview.appspot.com/31239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
Original cl description:
Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
- 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
- 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
- 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
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
