标签: apprtcdemo

我正在开发一个聊天应用程序并完成它.现在我也想实现视频聊天.经过研究很多,我决定选择"WebRTC"库.

我做了什么？

1)能够在本地服务器上运行AppRtcDemo,并且它在浏览器之间正常工作.

参考:http://www.webrtc.org/reference/getting-started

2)能够构建Android AppRtcDemo.But当我运行时说"Cross origin not support".

经过研究,我在webrtc讨论中发现,为了解决这个问题,我需要设置自己的转向服务器.

3)所以我安装了webrtc推荐的最新rfc5766TurnServer.我成功运行了转向服务器.

参考:http://code.google.com/p/rfc5766-turn-server/

我对ApprtcDemo(web)和(Android)进行了以下更改,以便与我的Turn服务器一起使用

1)apprtc.py

更换:

turn_url = 'https://computeengineondemand.appspot.com/'
turn_url = turn_url + 'turn?' + 'username=' + user + '&key=4080218913'

指向我的转弯服务器:

turn_url = 'http://192.168.5.85:3478/?service=turn&username=biraj'

2)index.html

更换:

var pcConfig = {{ pc_config|safe }};

附:

var pcConfig = {"iceServers": [{"url": "stun:stun.l.google.com:19302"},            {"url":"turn:biraj@192.168.5.85:3479", "credential":"0x5b04123c3eec4cf0be64ab909bb2ff5b"}]};

Android的

1)AppRTCDemoActivity.java

更换:

roomInput.setText("https://apprtc.appspot.com/?r=");

使用我的本地apprtc服务器:

roomInput.setText("http://192.168.5.86:8080/?r=");

2)AppRTCClient.java

在private PeerConnection.IceServer requestTurnServer(String url){}功能上

更换:

connection.addRequestProperty("origin", "https://apprtc.appspot.com");

附:

connection.addRequestProperty("origin", "http://192.168.5.86:8080");

3)/assets/channel.html

更换:

<script src="https://apprtc.appspot.com/_ah/channel/jsapi"></script>

附: …

我需要使用WebRTC for android将特定的裁剪(面部)视频发送到videoChannel.我能够操作CameraRession类WebRTC来裁剪面部.现在我将它设置为ImageView. listenForBytebufferFrames()的Camera1Session.java

private void listenForBytebufferFrames() {
    this.camera.setPreviewCallbackWithBuffer(new PreviewCallback() {
        public void onPreviewFrame(byte[] data, Camera callbackCamera) {
            Camera1Session.this.checkIsOnCameraThread();
            if(callbackCamera != Camera1Session.this.camera) {
                Logging.e("Camera1Session", "Callback from a different camera. This should never happen.");
            } else if(Camera1Session.this.state != Camera1Session.SessionState.RUNNING) {
                Logging.d("Camera1Session", "Bytebuffer frame captured but camera is no longer running.");
            } else {
                mFrameProcessor.setNextFrame(data, callbackCamera);
                long captureTimeNs = TimeUnit.MILLISECONDS.toNanos(SystemClock.elapsedRealtime());
                if(!Camera1Session.this.firstFrameReported) {
                    int startTimeMs = (int)TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - Camera1Session.this.constructionTimeNs);
                    Camera1Session.camera1StartTimeMsHistogram.addSample(startTimeMs);
                    Camera1Session.this.firstFrameReported = true;
                }

                ByteBuffer byteBuffer1 = ByteBuffer.wrap(data); …

我正在尝试将音频重定向到AppRTC iOS示例中的扬声器.

我试过了:

AVAudioSession* session = [AVAudioSession sharedInstance];

//error handling
BOOL success;
NSError* error;

//set the audioSession category. 
//Needs to be Record or PlayAndRecord to use audioRouteOverride:  

success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
                         error:&error];

if (!success)  NSLog(@"AVAudioSession error setting category:%@",error);

//set the audioSession override
success = [session overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker
                                      error:&error];
if (!success)  NSLog(@"AVAudioSession error overrideOutputAudioPort:%@",error);

//activate the audio session
success = [session setActive:YES error:&error];
if (!success) NSLog(@"AVAudioSession error activating: %@",error);
else NSLog(@"audioSession active");

没有错误,但它不起作用.我怎样才能解决这个问题？

我在android上使用APPRTCdemo应用程序.我试图让它播放来自其他同伴的声音,就像在Android设置中设置音量一样响亮.因此,如果用户将设备静音,则不会听到音频.我几乎尝试了所有Android API调用,但似乎没有任何影响.这些是我尝试的东西:AudioManager audioManager =(AudioManager)_context.getSystemService(Context.AUDIO_SERVICE); int volume = audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);

    volume = audioManager.getStreamVolume(AudioManager.STREAM_MUSIC);
    volume =audioManager.getStreamVolume(AudioManager.STREAM_RING);
    volume =audioManager.getStreamVolume(AudioManager.STREAM_ALARM);
    volume =audioManager.getStreamVolume(AudioManager.STREAM_NOTIFICATION);
    volume =audioManager.getStreamVolume(AudioManager.STREAM_SYSTEM);

    audioManager.setStreamVolume(AudioManager.STREAM_MUSIC, 0, AudioManager.FLAG_REMOVE_SOUND_AND_VIBRATE);
    audioManager.setStreamVolume(AudioManager.MODE_IN_COMMUNICATION, 0, AudioManager.FLAG_REMOVE_SOUND_AND_VIBRATE);
    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, 0,AudioManager.FLAG_REMOVE_SOUND_AND_VIBRATE);
    audioManager.setStreamMute(AudioManager.MODE_IN_COMMUNICATION, true);
    audioManager.setStreamMute(AudioManager.STREAM_SYSTEM, true);
    audioManager.setStreamMute(AudioManager.STREAM_RING, true);
    audioManager.setStreamMute(AudioManager.STREAM_MUSIC, true);
    audioManager.setStreamMute(AudioManager.STREAM_NOTIFICATION, true);

我正在将WebRTC本地实现到Android中.我能够编译和运行这里描述的代码http://www.webrtc.org/native-code/android,但我遇到了一个问题,apprtc.appspot.com显然没有按照假设返回一个通道令牌:

01-05 20:01:51.230  15488-15488/org.appspot.apprtc E/AppRTCDemoActivity? Fatal error: Missing channelToken in HTML: <!DOCTYPE html>
    <!--
    *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
    *
    *  Use of this source code is governed by a BSD-style license
    *  that can be found in the LICENSE file in the root of the source
    *  tree.
    -->
    <html>
    <head>
    <title>WebRTC Reference App</title>
    <meta charset="utf-8">
    <meta name="description" content="WebRTC reference app">
    <meta name="viewport" content="width=device-width, user-scalable=no, initial-scale=1, maximum-scale=1">
    <script type='text/javascript'>window.mod_pagespeed_start = Number(new …

最近在研究android中的webrtc。

我想单击SurfaceView A 使其全屏，而另一个SurfaceView B 变小，或者单击 B。但我发现很难在运行时更改两个不同 SurfaceView 的 z 顺序。

这是我的布局

<FrameLayout
android:layout_width="match_parent"
android:layout_height="match_parent"
app:layout_behavior="@string/appbar_scrolling_view_behavior"
tools:showIn="@layout/app_bar_main">

<org.webrtc.PercentFrameLayout
    android:id="@+id/remote_video_layout"
    android:layout_width="match_parent"
    android:layout_height="match_parent">

    <org.webrtc.SurfaceViewRenderer
        android:id="@+id/remote_video_view"
        android:layout_width="wrap_content"
        android:layout_height="wrap_content" />

</org.webrtc.PercentFrameLayout>

<org.webrtc.PercentFrameLayout
    android:id="@+id/local_video_layout"
    android:layout_width="match_parent"
    android:layout_height="match_parent">

    <org.webrtc.SurfaceViewRenderer
        android:id="@+id/local_video_view"
        android:layout_width="wrap_content"
        android:layout_height="wrap_content"
        android:keepScreenOn="true" />

</org.webrtc.PercentFrameLayout>

PercentFrameLayout 是 ViewGroup 的子类，用于测量屏幕百分比。
我尝试在运行时使用 setZOrderMediaOverlay 但失败了。我发现SDK说

请注意，必须在将表面视图的包含窗口附加到窗口管理器之前设置此项。