pkasting@chromium.org 16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
..
2013-12-12 16:55:37 +00:00
2013-12-12 16:55:37 +00:00

This directory contains a sample app for sending and receiving video and audio
on Android. It further lets you enable and disable some call quality
enhancements such as echo cancellation, noise suppression etc.

Prerequisites:
- Make sure gclient is checking out tools necessary to target Android: your
  .gclient file should contain a line like:
  target_os = ['android']
  Make sure to re-run gclient sync after adding this to download the tools.
- Env vars need to be set up to target Android; easiest way to do this is to run
  (from the libjingle trunk directory):
  . ./build/android/envsetup.sh
  Note that this clobbers any previously-set $GYP_DEFINES so it must be done
  before the next item.
- Set up webrtc-related GYP variables:
  export GYP_DEFINES="$GYP_DEFINES java_home=</path/to/JDK>"
- Finally, run "gclient runhooks" to generate Android-targeting .ninja files.

Example of building the app:
cd <path/to/repository>/trunk
ninja -C out/Debug WebRTCDemo

It can then be installed and run on the device:
adb install -r out/Debug/WebRTCDemo-debug.apk