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| author | Chih-hung Hsieh <chh@google.com> | 2015-12-01 17:07:48 +0000 |
|---|---|---|
| committer | android-build-merger <android-build-merger@google.com> | 2015-12-01 17:07:48 +0000 |
| commit | a4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch) | |
| tree | 672a185b294789cf991f385c3e395dd63bea9063 /webrtc/examples/android/media_demo/README | |
| parent | 3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff) | |
| parent | fe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff) | |
| download | webrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz | |
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579
* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
WIP: Changes after merge commit 'cb3f9bd'
Make the nonlinear beamformer steerable
Utilize bitrate above codec max to protect video.
Enable VP9 internal resize by default.
Filter overlapping RTP header extensions.
Make VCMEncodedFrameCallback const.
MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
Remove redudant encoder rate calls.
Create isolate files for nonparallel tests.
Register header extensions in RtpRtcpObserver to avoid log spam.
Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
ACM: Move NACK functionality inside NetEq
Fix chromium-style warnings in webrtc/sound/.
Create a 'webrtc_nonparallel_tests' target.
Update scalability structure data according to updates in the RTP payload profile.
audio_coding: rename interface -> include
Rewrote perform_action_on_all_files to be parallell.
Update reference indices according to updates in the RTP payload profile.
Disable P2PTransport...TestFailoverControlledSide on Memcheck
pass clangcl compile options to ignore warnings in gflags.cc
...
Diffstat (limited to 'webrtc/examples/android/media_demo/README')
| -rw-r--r-- | webrtc/examples/android/media_demo/README | 24 |
1 files changed, 24 insertions, 0 deletions
diff --git a/webrtc/examples/android/media_demo/README b/webrtc/examples/android/media_demo/README new file mode 100644 index 0000000000..af71f38f46 --- /dev/null +++ b/webrtc/examples/android/media_demo/README @@ -0,0 +1,24 @@ +This directory contains a sample app for sending and receiving 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 |