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webrtc-android's Introduction

This Android project is a mirror of the Android AppRTC demo found in the webrtc project. However, this Android application includes the sources for the libjingle_peerconnection library normally included pre-compiled with the Android AppRTC demo. Note that the sources in this project are not in any way ours. We are just mirroring what is all ready available to developers in an Android project form so developers can tweak all the java sources without building or downloading the WebRTC project

Development

Working with this project requires use of the Maven Android SDK Deployer. Follow instructions there before proceeding with development. Once you have installed your Android SDK to your local maven repository you can compile and modify the project locally.

The native dependency used is pre-compiled by us and available on maven central. Note that any library and application changes are bound by the libjingle_peerconnection.so file available. However, feel free to build your own .so file and drop it in this project.

Modifications

This repository is meant to be a starting point for exploration with WebRTC on Android. Only configuration based changes will be considered for this repository. The sources for the application and library will not be different from what is available within the WebRTC project. If you have some changes for the library or application that you think are relevant, feel free to test them out with this project, but please contribute to the official WebRTC repository and the changes will be incorporated here.

Updates

We won't make any promises on periodic updates to this repository, but rest assured we interact with WebRTC on a daily basis and if any noteworthy updates become available we will update the project shortly thereafter.

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webrtc-android's Issues

I'm getting "Incorrect strides" all time and video isn't play

I'm getting the following log when I stablish a videocall, audio works but video isn't. If I do the try with two Nexus 5 it works :S!!

02-04 18:42:30.392  26474-26694/com.prueberia.webrtctests I/libjingle﹕ webrtc: (vie_channel.cc:1763): OnInitializeDecoder 100 VP8
02-04 18:42:30.412  26474-26763/com.prueberia.webrtctests I/libjingle﹕ WebRtcRenderAdapter (channel 0) frame size changed to: 480x640
02-04 18:42:30.412  26474-26763/com.prueberia.webrtctests D/VideoRendererGui﹕ ID: 0. YuvImageRenderer.setSize: 480 x 640
02-04 18:42:30.412  26474-26763/com.prueberia.webrtctests E/VideoRendererGui﹕ Incorrect strides 544, 272, 272
[..continues to the infinite..]

I'm so confused, anybody can bring few light on it ?

Why is WebRTC video file so big?

I'm trying to save video of WebRTC call. It's working but the video file is ridiculously big.

val recording = VideoFileRenderer(mPath, mWidth, mHeight,  
rootEglBase.eglBaseContext)
localVideoTrack.addSink(recording)

I think it's obvious that WebRTC doesn't apply any kind of compression to the video. Even though. I'm setting DefaultVideoEncoderFactory costructor's VP8 encoder to true.

val defaultVideoEncoderFactory = DefaultVideoEncoderFactory(rootEglBase.eglBaseContext, true, true)

Crash happens while video/audio call is on going

Hi,
The following crash is occured when i made a audio/video call.Can any one suggest me a solution for this
java.lang.RuntimeException: Error during updateTexImage (see logcat for details)
at android.graphics.SurfaceTexture.nativeUpdateTexImage(Native Method)
at android.graphics.SurfaceTexture.updateTexImage(SurfaceTexture.java:244)
at org.webrtc.SurfaceTextureHelper.updateTexImage(SurfaceTextureHelper.java:244)
at org.webrtc.SurfaceTextureHelper.tryDeliverTextureFrame(SurfaceTextureHelper.java:258)
at org.webrtc.SurfaceTextureHelper.access$500(SurfaceTextureHelper.java:36)
at org.webrtc.SurfaceTextureHelper$3.onFrameAvailable(SurfaceTextureHelper.java:132)
at android.graphics.SurfaceTexture$1.handleMessage(SurfaceTexture.java:207)
at android.os.Handler.dispatchMessage(Handler.java:110)
at android.os.Looper.loop(Looper.java:203)
at android.os.HandlerThread.run(HandlerThread.java:61)

Custom audio device / source

I am developing native Android WebRTC client that is supposed to stream audio from a custom device (I am getting audio stream via Bluetooth from that device). Is possible to hook up custom audio stream?

mute audio

Can anybody tell how to mute the audio in android webrtc , i think using mediastream , but how can we do

Complie

Hello,
I am trying to use your code but i am not being able to compile your code so can you please let us know how to compile this.

How I can take screenshot of WebRTC View?

I used this library in here : http://mvnrepository.com/artifact/io.pristine/libjingle/9944
and use the gl.glReadPixels(x, 0, w, y+h, GL10.GL_RGBA, GL10.GL_UNSIGNED_BYTE, intBuffer);
method in my Activity,but returned bmp is black(perhaps because not completely worked method OnDraw).And I do not can extended my custom class from VideoRendererGui class,and redefine OnDraw method.
update: I am use old example https://libjingle.googlecode.com/svn-history/r287/trunk/talk/examples/android/ but in my version libjingle(9944) does not exist the "deepcopy" method in I420Frame
and my View is green,and after 5-10 seconds appears "Camera failure" error.
Can you help me?

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