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  • Camera Preview App in Android throwing many errors (Nexus 4)

    - by Jagatheesan Jack
    I am trying to develop a camera app that takes a picture and saves it in a SQLite database. I get a lot of errors when executing the application. My code is as below. Any idea? CameraActivity.java private Camera mCamera; private CameraPreview mPreview; private int CAMERA_RETURN_CODE=100; private static final String TAG = "Take_Picture"; public static final int MEDIA_TYPE_IMAGE = 1; public static final int MEDIA_TYPE_VIDEO = 2; private Bitmap cameraBmp; private int MAX_FACES = 1; private Face[] faceList; public RectF[] rects; private Canvas canvas; private Drawable pictureDataDrawable; private MySQLiteHelper database; @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.camera_activity); //this.requestWindowFeature(Window.FEATURE_NO_TITLE); //Create an instance of Camera mCamera = getCameraInstance(); setCameraDisplayOrientation(this, 0, mCamera); // Create our Preview view and set it as the content of our activity. mPreview = new CameraPreview(this, mCamera); FrameLayout preview = (FrameLayout) findViewById(R.id.camera_preview); preview.addView(mPreview); database = new MySQLiteHelper(getApplicationContext()); Button captureButton = (Button) findViewById(R.id.button_capture); captureButton.setOnClickListener( new View.OnClickListener() { private PictureCallback mPicture; @Override public void onClick(View v) { //mCamera.startPreview(); // get an image from the camera mCamera.takePicture(null, null, mPicture); PictureCallback mPicture = new PictureCallback() { @Override public void onPictureTaken(byte[] data, Camera camera) { try{ if (data != null) database.addEntry(data); //mCamera.startPreview(); } catch(Exception e){ Log.d(TAG, e.getMessage()); } } } ); } /** A safe way to get an instance of the Camera object. */ public static Camera getCameraInstance(){ Camera c = null; try { c = Camera.open(c.getNumberOfCameras()-1); // attempt to get a Camera instance } catch (Exception e){ // Camera is not available (in use or does not exist) } return c; // returns null if camera is unavailable } public static void setCameraDisplayOrientation(Activity activity, int cameraId, android.hardware.Camera camera) { android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo(); android.hardware.Camera.getCameraInfo(cameraId, info); int rotation = activity.getWindowManager().getDefaultDisplay() .getRotation(); int degrees = 360; /*switch (rotation) { case Surface.ROTATION_0: degrees = 0; break; case Surface.ROTATION_90: degrees = 90; break; case Surface.ROTATION_180: degrees = 180; break; case Surface.ROTATION_270: degrees = 270; break; }*/ int result; if (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) { result = (info.orientation + degrees) % 360; result = (360 - result) % 360; // compensate the mirror } else { // back-facing result = (info.orientation - degrees + 360) % 360; } camera.setDisplayOrientation(result); } @Override protected void onPause() { super.onPause(); //releaseMediaRecorder(); // if you are using MediaRecorder, release it first releaseCamera(); // release the camera immediately on pause event } private void releaseCamera(){ if (mCamera != null){ mCamera.release(); // release the camera for other applications mCamera = null; } } public void startFaceDetection(){ // Try starting Face Detection Camera.Parameters params = mCamera.getParameters(); // start face detection only *after* preview has started if (params.getMaxNumDetectedFaces() > 0){ // camera supports face detection, so can start it: mCamera.startFaceDetection(); } } CameraPreview.java public class CameraPreview extends SurfaceView implements SurfaceHolder.Callback { private SurfaceHolder mHolder; private Camera mCamera; private String TAG; private List<Size> mSupportedPreviewSizes; public CameraPreview(Context context, Camera camera) { super(context); mCamera = camera; // Install a SurfaceHolder.Callback so we get notified when the // underlying surface is created and destroyed. mHolder = getHolder(); mHolder.addCallback(this); // deprecated setting, but required on Android versions prior to 3.0 mHolder.setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS); } public void surfaceCreated(SurfaceHolder holder) { // The Surface has been created, now tell the camera where to draw the preview. try { mCamera.setPreviewDisplay(holder); mCamera.setDisplayOrientation(90); mCamera.startPreview(); } catch (IOException e) { Log.d(TAG, "Error setting camera preview: " + e.getMessage()); } } public void surfaceDestroyed(SurfaceHolder holder) { // empty. Take care of releasing the Camera preview in your activity. } public void surfaceChanged(SurfaceHolder holder, int format, int w, int h) { // If your preview can change or rotate, take care of those events here. // Make sure to stop the preview before resizing or reformatting it. if (mHolder.getSurface() == null){ // preview surface does not exist return; } // stop preview before making changes try { mCamera.stopPreview(); } catch (Exception e){ // ignore: tried to stop a non-existent preview } try { mCamera.setPreviewDisplay(mHolder); mCamera.startPreview(); } catch (Exception e){ Log.d(TAG, "Error starting camera preview: " + e.getMessage()); } } public void setCamera(Camera camera) { if (mCamera == camera) { return; } mCamera = camera; if (mCamera != null) { List<Size> localSizes = mCamera.getParameters().getSupportedPreviewSizes(); mSupportedPreviewSizes = localSizes; requestLayout(); try { mCamera.setPreviewDisplay(mHolder); } catch (IOException e) { e.printStackTrace(); } /* Important: Call startPreview() to start updating the preview surface. Preview must be started before you can take a picture. */ mCamera.startPreview(); } } MySQLiteHelper.java private static final int count = 0; public static final String TABLE_IMAGE = "images"; public static final String COLUMN_ID = "_id"; public static final String PICTURE_DATA = "picture"; public static final String DATABASE_NAME = "images.db"; public static final int DATABASE_VERSION = 1; public static final String DATABASE_CREATE = "create table " + TABLE_IMAGE + "(" + COLUMN_ID + " integer primary key autoincrement, " + PICTURE_DATA + " blob not null);"; public static SQLiteDatabase database; private static String TAG = "test"; public MySQLiteHelper(Context context) { super(context, DATABASE_NAME, null, DATABASE_VERSION); // TODO Auto-generated constructor stub } public MySQLiteHelper(Context context, String name, CursorFactory factory, int version, DatabaseErrorHandler errorHandler) { super(context, name, factory, version, errorHandler); // TODO Auto-generated constructor stub } @Override public void onCreate(SQLiteDatabase database) { database.execSQL(DATABASE_CREATE); } @Override public void onUpgrade(SQLiteDatabase db, int oldVersion, int newVersion) { Log.w(MySQLiteHelper.class.getName(), "Upgrading database from version " + oldVersion + " to " + newVersion + ", which will destroy all old data"); db.execSQL("DROP TABLE IF EXISTS " + TABLE_IMAGE); onCreate(db); } /** * @param args */ public static void main(String[] args) { // TODO Auto-generated method stub } public void addEntry(byte [] array) throws SQLiteException{ ContentValues cv = new ContentValues(); //cv.put(KEY_NAME, name); cv.put(PICTURE_DATA, array); database.insert( TABLE_IMAGE, null, cv ); Log.w(TAG , "added " +count+ "images"); database.close(); } Errors 11-07 23:28:39.050: E/mm-libcamera2(176): PROFILE HAL: stopPreview(): E: 1383838119.067589459 11-07 23:28:39.050: E/mm-camera(201): config_MSG_ID_STOP_ACK: streamon_mask is not clear. Should not call PP_Release_HW 11-07 23:28:39.090: E/QCameraHWI(176): android::status_t android::QCameraHardwareInterface::setPreviewWindow(preview_stream_ops_t*):Received Setting NULL preview window 11-07 23:28:39.090: E/QCameraHWI(176): android::status_t android::QCameraHardwareInterface::setPreviewWindow(preview_stream_ops_t*): mPreviewWindow = 0x0x0, mStreamDisplay = 0x0xb8a9df90 11-07 23:28:39.090: E/mm-camera(201): config_shutdown_pp Camera not in streaming mode. Returning. 11-07 23:28:39.090: E/mm-camera(201): vfe_ops_deinit: E 11-07 23:28:39.120: E/qcom_sensors_hal(533): hal_process_report_ind: Bad item quality: 11 11-07 23:28:39.310: E/qcom_sensors_hal(533): hal_process_report_ind: Bad item quality: 11 11-07 23:28:39.330: E/mm-camera(201): sensor_load_chromatix: libchromatix_imx119_preview.so: 30 11-07 23:28:39.340: E/mm-camera(201): vfe_ops_init: E 11-07 23:28:39.360: E/mm-camera(201): vfe_legacy_stats_buffer_init: AEC_STATS_BUFNUM 11-07 23:28:39.360: E/mm-camera(201): vfe_legacy_stats_buffer_init: AEC_STATS_BUFNUM 11-07 23:28:39.360: E/mm-camera(201): mctl_init_stats_proc_info: snap_max_line_cnt =25776 11-07 23:28:39.440: E/QCameraHWI(176): android::status_t android::QCameraHardwareInterface::setPreviewWindow(preview_stream_ops_t*): mPreviewWindow = 0x0xb8aa1780, mStreamDisplay = 0x0xb8a9df90 11-07 23:28:39.440: E/mm-camera(201): config_proc_CAMERA_SET_INFORM_STARTPREVIEW 11-07 23:28:39.450: E/mm-camera(201): config_update_stream_info Storing stream parameters for video inst 1 as : width = 640, height 480, format = 1 inst_handle = 810081 cid = 0 11-07 23:28:39.490: E/mm-camera(201): config_update_stream_info Storing stream parameters for video inst 3 as : width = 640, height 480, format = 1 inst_handle = 830083 cid = 0 11-07 23:28:39.490: E/mm-camera(201): config_update_stream_info Storing stream parameters for video inst 4 as : width = 512, height 384, format = 1 inst_handle = 840084 cid = 0 11-07 23:28:39.500: E/mm-camera(201): config_decide_vfe_outputs: Ports Used 3, Op mode 1 11-07 23:28:39.500: E/mm-camera(201): config_decide_vfe_outputs Current mode 0 Full size streaming : Disabled 11-07 23:28:39.500: E/mm-camera(201): config_decide_vfe_outputs: Primary: 640x480, extra_pad: 0x0, Fmt: 1, Type: 1, Path: 1 11-07 23:28:39.500: E/mm-camera(201): config_decide_vfe_outputs: Secondary: 640x480, extra_pad: 0x0, Fmt: 1, Type: 3, Path: 4 11-07 23:28:39.510: E/mm-camera(201): config_update_inst_handles Updated the inst handles as 810081, 830083, 0, 0 11-07 23:28:39.631: E/mm-camera(201): sensor_load_chromatix: libchromatix_imx119_preview.so: 30 11-07 23:28:39.631: E/mm-camera(201): camif_client_set_params: camif has associated with obj mask 0x1 11-07 23:28:39.631: E/mm-camera(201): config_v2_CAMERA_START_common CAMIF_PARAMS_ADD_OBJ_ID failed -1 11-07 23:28:39.641: E/mm-camera(201): vfe_operation_config: format 3 11-07 23:28:39.641: E/mm-camera(201): vfe_operation_config:vfe_op_mode=5 11-07 23:28:39.641: E/mm-camera(201): Invalid ASD Set Params Type 11-07 23:28:39.641: E/mm-camera(201): vfe_set_bestshot: Bestshot mode not changed

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  • AVFoundation: Video to OpenGL texture working - How to play and sync audio?

    - by j00hi
    I've managed to load a video-track of a movie frame by frame into a OpenGL texture with AVFoundation. I followed the steps described in the answer here: iOS4: how do I use video file as an OpenGL texture? and took some code from the GLVideoFrame sample from WWDC2010 which can be downloaded here: http://bit.ly/cEf0rM How do I play the audio-track of the movie synchronously to the video. I think it would not be a good idea to play it in a separate player, but to use the audio-track of the same AVAsset. AVAssetTrack* audioTrack = [[asset tracksWithMediaType:AVMediaTypeAudio] objectAtIndex:0]; I retrieve a videoframe and it's timestamp in the CADisplayLink-callback via CMSampleBufferRef sampleBuffer = [self.readerOutput copyNextSampleBuffer]; CMTime timestamp = CMSampleBufferGetPresentationTimeStamp( sampleBuffer ); where readerOutput is of type AVAssetReaderTrackOutput* How to get the corresponding audio-samples? And how to play them? Edit: I've looked around a bit and I think, best would be to use AudioQueue from the AudioToolbox.framework using the approach described here: AVAssetReader and Audio Queue streaming problem There is also an audio-player in the AVFoundation: AVAudioPlayer. But I don't know exactly how I should pass data to it's initWithData-initializer which expects NSData. Furthermore I don't think it's the best choice for my case because a new AVAudioPlayer-instance would have to be created for every new chunk of audio samples, as I understand it. Any other suggestions? What's the best way to play the raw audio samples which i get from the AVAssetReaderTrackOutput?

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  • WCF - CODEGEN: Generating message contract since message FileRequest has headers

    - by Tri Q
    I am aware that there is a similar question here with no solution. I'm working on a WCF streaming service over HTTP. Here are my MessageContract [MessageContract] public class FileRequest { #region Message Header [MessageHeader(MustUnderstand = true)] public Credential Credentials { get; set; } #endregion #region Message body [MessageBodyMember(Order = 1)] public FileInfo FileInfo { get; set; } #endregion #region Ctor // ... #endregion } [MessageContract] public class FileRequestResponse { #region Message Header [MessageHeader(MustUnderstand = true)] public FileInfo FileHeader { get; set; } [MessageHeader(MustUnderstand = true)] public OperationResult<bool> OperationResult { get; set; } #endregion #region Message Body [MessageBodyMember] public Stream FileStream { get; set; } #endregion #region Constructor // ... #endregion } Here is my ServiceContract [ServiceContract(Namespace = "https://service.contract.example.com")] public interface IUpdateService { [OperationContract(Action = "GetUpdates")] OperationResult<List<FileInfo>> GetUpates(ApplicationInfo applicationInfo, Credential credential); [OperationContract(Action = "GetFile")] FileRequestResponse FileRequest(FileRequest fileRequest); } Now the question is why I am getting this error: // CODEGEN: Generating message contract since message FileRequest has headers When I add my service reference. The end result is that the service contract wraps the FileRequest operation into a wrapper which I do not want. public FileInfo FileRequest(Credential Credentials, FileInfo, out OperationResult<bool> OperationResult, out System.IO.Stream FileStream) NOTE: I have not checked the "Always generate message contracts" in the service reference.

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  • TimoutException occurs over a network but not locally

    - by Gibsnag
    I have a program with three WCF services and when I run them locally (i.e: Server and Clients are all on localhost) everything works. However when I test them across a network I get a TimoutException on two services but not the other. I've disabled the firewalls on all the machines involved in the test. I can both ping the server and access the wsdl "You have created a service" webpage from the client The service that works uses a BasicHttpBinding with streaming and the two which don't work use WSDualHttpBinding. The Services that use WSDualHttpBinding both have CallbackContracts. I apologise for the vagueness of this question but I'm not really sure what code to include or where to even start looking for the solution to this. Non-working bindings: public static Binding CreateHTTPBinding() { var binding = new WSDualHttpBinding(); binding.MessageEncoding = WSMessageEncoding.Mtom; binding.MaxBufferPoolSize = 2147483647; binding.MaxReceivedMessageSize = 2147483647; binding.Security.Mode = WSDualHttpSecurityMode.None; return binding; } Exception Stack Trace: Unhandled Exception: System.TimeoutException: The open operation did not complete within the allotted timeout of 00:01:00. The time allotted to this operation may have been a portion of a longer timeout. Server stack trace: at System.ServiceModel.Channels.ReliableRequestor.ThrowTimeoutException() at System.ServiceModel.Channels.ReliableRequestor.Request(TimeSpan timeout) at System.ServiceModel.Channels.ClientReliableSession.Open(TimeSpan timeout) at System.ServiceModel.Channels.ClientReliableDuplexSessionChannel.OnOpen(TimeSpan timeout) at System.ServiceModel.Channels.CommunicationObject.Open(TimeSpan timeout) at System.ServiceModel.Channels.ServiceChannel.OnOpen(TimeSpan timeout) at System.ServiceModel.Channels.CommunicationObject.Open(TimeSpan timeout) at System.ServiceModel.Channels.ServiceChannel.CallOpenOnce.System.ServiceModel.Channels.ServiceChannel.ICallOnce.Call(ServiceChannel channel, TimeSpan timeout) at System.ServiceModel.Channels.ServiceChannel.CallOnceManager.CallOnce(TimeSpan timeout, CallOnceManager cascade) at System.ServiceModel.Channels.ServiceChannel.EnsureOpened(TimeSpan timeout) at System.ServiceModel.Channels.ServiceChannel.Call(String action, Boolean oneway, ProxyOperationRuntime operation, Object[] ins, Object[] outs, TimeSpan timeout) at System.ServiceModel.Channels.ServiceChannel.Call(String action, Boolean oneway, ProxyOperationRuntime operation, Object[] ins, Object[] outs) at System.ServiceModel.Channels.ServiceChannelProxy.InvokeService(IMethodCallMessage methodCall, ProxyOperationRuntime operation) at System.ServiceModel.Channels.ServiceChannelProxy.Invoke(IMessage message) Exception rethrown at [0]: at System.Runtime.Remoting.Proxies.RealProxy.HandleReturnMessage(IMessage reqMsg, IMessage retMsg) at System.Runtime.Remoting.Proxies.RealProxy.PrivateInvoke(MessageData& msgData, Int32 type) at IDemeService.Register() at DemeServiceClient.Register() at DemeClient.Client.Start() at DemeClient.Program.Main(String[] args)

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  • Where exactly should i add crossdomain.xml file?

    - by SpikETidE
    Hi everyone... i am trying to create a internet radio.... I use icecast2 for streaming..... edcast plugin with winamp to send the music to icecast... and the xspf web music player (http://musicplayer.sourceforge.net/) to connect the user to the icecast server and play the music.... The setup works great and i can broadcast and receive on the local network i use to test the radio.. using xampp... Now the icecast broadcasts online from a windows server with the ip address say xx.xx.xxx.xxx The webpage in which the flash player is embedded is uploaded to www.xyz.com/images/radio This domain has the same ip address from where the icecast server runs. Now when i run the webpage to connect to the radio with the flash player, i get the error in firebug as "xx.xx.xxx.xxx:8000/crossdomain.xml 404 NOT FOUND" But i have created a crossdomain.xml file in the root of the xx.xx.xxx.xxx server... Still it doesn't recognize the file... Can anyone tell me where exactly i should create the file for my setting...??? Thanks a lot in advance.....

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  • FMS NetConnection.Connect.Close happening when starts and even in the middle of video in Flash with

    - by Sunil Kumar
    Hi I have developed a Flash Video player in Flash CS3 with Action Script 2.0 to play video from Adobe Flash Media Server 3.5. To play video from FMS 3.5, first I have to verify my swf file on FMS 3.5 server console so that it can be ensure that RTMP video URL only be play in verified SWF file. Right now I am facing problem of "NetConnection.Connect.Close" when I try to connect my NetConnection Object to FMS 3.5 to stream video from that server. So now I am getting this message "NetConnection.Connect.Close" from FMS 3.5. When this is happening in my office area at the same time when I am checking the the same video url from out side the office (With help of my friends who is in another office) area it is working fine. My friends naver faced even a single issue with NetConnection.Connect.Close. But in my office when I got message NetConnection.Connect.Close, I can play another streaming video very well like mtv.com jaman.com rajshri.com etc. Some time FMS works fine and video starts playing but in the middle of the video same thing happen "NetConnection.Connect.Close" There is no issue of Bandwidth in my office. I do't know why this is happening. Please see the message when I am getting "NetConnection.Connect.Close" message. NetConn == data: NetConn == objectEncoding: 0 NetConn == description: Connection succeeded. NetConn == code: NetConnection.Connect.Success NetConn == level: status NetConn == level: status NetConn == code: NetConnection.Connect.Closed Please help Thanks & regards Sunil Kumar

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  • C++ JSON parser

    - by pollux
    Dear reader, I'm working on a twitter client which uses the twitter streaming json api. Twitter advices JSON as XML version is deprecated. I'm looking for a good JSON parser which can parse the json data below. I'm receiving this JSON which I want to be able to read/parse using a JSON parser. { "in_reply_to_status_id": null, "text": "Home-plate umpire Crawford gets stung http://tinyurl.com/27ujc86", "favorited": false, "coordinates": null, "in_reply_to_user_id": null, "source": "<a href=\"http://apiwiki.twitter.com/\" rel=\"nofollow\">API</a>", "geo": null, "created_at": "Fri Jun 18 15:12:06 +0000 2010", "place": null, "user": { "profile_text_color": "333333", "screen_name": "HostingViral", "time_zone": "Pacific Time (US & Canada)", "url": "http://bit.ly/1Way7P", "profile_link_color": "228235", "profile_background_image_url": "http://s.twimg.com/a/1276654401/images/themes/theme14/bg.gif", "description": "Full time Internet Marketer - Helping other reach their Goals\r\nhttp://wavemarker.com", "statuses_count": 1944, "profile_sidebar_fill_color": "c7b7c7", "profile_background_tile": true, "contributors_enabled": false, "lang": "en", "notifications": null, "created_at": "Wed Dec 30 07:50:52 +0000 2009", "profile_sidebar_border_color": "120412", "following": null, "geo_enabled": false, "followers_count": 2485, "protected": false, "friends_count": 2495, "location": "Working at Home", "name": "Johnathan Thomas", "verified": false, "profile_background_color": "131516", "profile_image_url": "http://a1.twimg.com/profile_images/600114776/nessykalvo421_normal.jpg", "id": 100439873, "utc_offset": -28800, "favourites_count": 0 }, "in_reply_to_screen_name": null, "id": 16477056501, "contributors": null, "truncated": false } *This is the raw string (above it beautified) * {"in_reply_to_status_id":null,"text":"Home-plate umpire Crawford gets stung http://tinyurl.com/27ujc86","favorited":false,"coordinates":null,"in_reply_to_user_id":null,"source":"<a href=\"http://apiwiki.twitter.com/\" rel=\"nofollow\">API</a>","geo":null,"created_at":"Fri Jun 18 15:12:06 +0000 2010","place":null,"user":{"profile_text_color":"333333","screen_name":"HostingViral","time_zone":"Pacific Time (US & Canada)","url":"http://bit.ly/1Way7P","profile_link_color":"228235","profile_background_image_url":"http://s.twimg.com/a/1276654401/images/themes/theme14/bg.gif","description":"Full time Internet Marketer - Helping other reach their Goals\r\nhttp://wavemarker.com","statuses_count":1944,"profile_sidebar_fill_color":"c7b7c7","profile_background_tile":true,"contributors_enabled":false,"lang":"en","notifications":null,"created_at":"Wed Dec 30 07:50:52 +0000 2009","profile_sidebar_border_color":"120412","following":null,"geo_enabled":false,"followers_count":2485,"protected":false,"friends_count":2495,"location":"Working at Home","name":"Johnathan Thomas","verified":false,"profile_background_color":"131516","profile_image_url":"http://a1.twimg.com/profile_images/600114776/nessykalvo421_normal.jpg","id":100439873,"utc_offset":-28800,"favourites_count":0},"in_reply_to_screen_name":null,"id":16477056501,"contributors":null,"truncated":false} I've tried multiple JSON parsers from json.org though I've tried 4 now and can't find one which can parse above json. Kind regards, Pollux

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  • How do I play back a WAV in ActionScript?

    - by Jeremy White
    Please see the class I have created at http://textsnip.com/51013f for parsing a WAVE file in ActionScript 3.0. This class is correctly pulling apart info from the file header & fmt chunks, isolating the data chunk, and creating a new ByteArray to store the data chunk. It takes in an uncompressed WAVE file with a format tag of 1. The WAVE file is embedded into my SWF with the following Flex embed tag: [Embed(source="some_sound.wav", mimeType="application/octet-stream")] public var sound_class:Class; public var wave:WaveFile = new WaveFile(new sound_class()); After the data chunk is separated, the class attempts to make a Sound object that can stream the samples from the data chunk. I'm having issues with the streaming process, probably because I'm not good at math and don't really know what's happening with the bits/bytes, etc. Here are the two documents I'm using as a reference for the WAVE file format: http://www.lightlink.com/tjweber/StripWav/Canon.html https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ Right now, the file IS playing back! In real time, even! But...the sound is really distorted. What's going on?

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  • What is the best way to send structs containing enum values via sockets in C.

    - by Axel
    I've lots of different structs containing enum members that I have to transmit via TCP/IP. While the communication endpoints are on different operating systems (Windows XP and Linux) meaning different compilers (gcc 4.x.x and MSVC 2008) both program parts share the same header files with type declarations. For performance reasons, the structures should be transmitted directly (see code sample below) without expensively serializing or streaming the members inside. So the question is how to ensure that both compilers use the same internal memory representation for the enumeration members (i.e. both use 32-bit unsigned integers). Or if there is a better way to solve this problem... //type and enum declaration typedef enum { A = 1, B = 2, C = 3 } eParameter; typedef enum { READY = 400, RUNNING = 401, BLOCKED = 402 FINISHED = 403 } eState; #pragma pack(push,1) typedef struct { eParameter mParameter; eState mState; int32_t miSomeValue; uint8_t miAnotherValue; ... } tStateMessage; #pragma pack(pop) //... send via socket tStateMessage msg; send(iSocketFD,(void*)(&msg),sizeof(tStateMessage)); //... receive message on the other side tStateMessage msg_received; recv(iSocketFD,(void*)(&msg_received),sizeof(tStateMessage)); Additionally... Since both endpoints are little endian maschines, endianess is not a problem here. And the pack #pragma solves alignment issues satisfactorily. Thx for your answers, Axel

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  • General question about DirectShow.NET, DirectShow and Windows Media Format

    - by Paul Andrews
    I searched and googled for an answer but couldn't find one. Basically I'm developing a webcam/audio streaming application which should capture audio and video from a pc (usb webcam/microphone) and send them to a receiving server. What the server will do with that it's another story and phase two (which I'm skipping for now) I wrote some code using DirectShow and Windows Media Format and it worked great for capture audio/video and sending them to another client, but there's a major problem: latency. Everywhere in the internet everyone gave me the same answer: "sorry dude but media format isn't for video conferencing, their codecs have too high latency". I thought I could skip the .wmv problems but seems like it's not possible to do... this road ends here then. So I saw a few examples with DirectShow.NET which were faster for both audio and video.. my question is: how come that DirectShow.NET is faster and better for video/audio conferencing? Shouldn't it be just a .NET porting of C++'s DirectShow? Am I missing something? I'm a bit confused at this point

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  • How does GMail implement Comet?

    - by Morgan Cheng
    With the help of HttpWatch, I tried to figure out how GMail implement Comet. I Login in GMail with two account, one in IE and the other in Firefox. Chatting in GTalk in GMail with some magic words like "WASSUP". Then, I logoff both GMail accounts, filter any http content without "WASSUP" string. The result shows which HTTP request is the streaming channel. (Note: I have to logoff. Otherwise, never-ending HTTP would not show content in HttpWatch.) The result is interesting. The URL for stream channel is like: https://mail/channel/bind?VER=8&at=xn3j33vcvk39lkfq..... There is no surprise that GMail do Comet in IE with IFRAME. The Http content starts with " Originally, I guessed that GMail do Comet in Firefox with multipart XmlHttpRequest. To my surprise, the response header doesn't have "multipart/x-mixed-replace" header. The response headers are as below: HTTP/1.1 200 OK Content-Type: text/plain; charset=utf-8 Cache-Control: no-cache, no-store, max-age=0, must-revalidate Pragma: no-cache Expires: Fri, 01 Jan 1990 00:00:00 GMT Date: Sat, 20 Mar 2010 01:52:39 GMT X-Frame-Options: ALLOWALL Transfer-Encoding: chunked X-Content-Type-Options: nosniff Server: GSE X-XSS-Protection: 0 Unfortunately, the HttpWatch doesn't tell whether a HTTP request is from XmlHttpRequest or not. The content is not HTML but JSON. It looks like a response for XHR, but that would not work for Comet without multipart/x-mixed-replace, right? Is there any way else to figure out how GMail implement Comet? Thanks.

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  • Problems with MediaPlayer, raw resources, stop and start

    - by arakn0
    Hello everybody, I'm new in Android development and I have the next question/problem. I'm playing around with the MediaPlayer class to reproduce some sounds/music. I am playing raw resources (res/raw) and it looks kind of easy. To play a raw resource, the MediaPlayer has to be initialized like this: MediaPlayer mp = MediaPlayer.create(appContext, R.raw.song); mp.start(); Until here there is no problem. The sound is played, and everything works fine. My problem appears when I want to add more options to my application. Specifically when I add the "Stop" button/option. Basically, what I want to do is...when I press "Stop", the music stops. And when I press "Start", the song/sound starts over. (pretty basic!) To stop the media player, you only have to call stop(). But to play the sound again, the media player has to be reseted and prepared. mp.reset(); mp.setDataSource(params); mp.prepare(); The problem is that the method setDataSource() only accepts as params a file path, Content Provider URI, streaming media URL path, or File Descriptor. So, since this method doesn't accept a resource identifier, I don't know how to set the data source in order to call prepare(). In addition, I don't understand why you can't use a Resouce identifier to set the data source, but you can use a resource identifier when initializing the MediaPlayer. I guess that I'm missing something. I wonder if I am mixing concepts, and the method stop() doesn't have to be called in the "Stop" button. Any help? Thanks in advanced!!!

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  • Frame Accurate Browser Launchable Video Player ... ?

    - by cliftonc
    I have a requirement where I need to enable playback (full screen) of a h.264 MPEG4 (thanks for the correction!) video from a local network, launchable from a browser link on a Windows workstation, and be frame accurate. By frame accurate I mean that I need to be able to scrub through the video in the same way you would with a vtr, stop at a frame, and then move backwards and forwards frame by frame (it is for a very specific compliance requirement where have to be able to check every frame if there is something that is potentially against broadcasting guidelines). The application itself is used to capture notes while viewing the material, so the end model is for a dual monitor workstation, with a web form in one, the video playing full screen in the second (no issue launching the video and manually having to move it to the second screen), and then the user controls the video via keyboard shortcuts or a jog shuttle. I have looked at QT, but the java bindings seem to be dead or nearly so, flash isn't frame accurate, VLC given its streaming heritage seems to be only able to move forward by a frame and not backwards, and all I have left are commercial offerings that in my experience are difficult and expensive to change. Any ideas of where I should look or alternative options? Any advice appreciated!

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  • Keymap issues with NX from Mac OS X Lion

    - by Andy
    I tried to answer the question from Mark: Keymap issues with NX from Mac OS X Lion to Ubuntu However, it is locked so I figured I would post a new question / answer. I have been trying to answer this for a few days now because I have no issues when connecting through NX Client (technically OpenNX) to FreeNX server from an iMac (with Lion), but if I try to connect with a Macbook Pro I get horrible keyboard binding issues. The fix that is working for me is to go into: ~/.nx/config/HOST.nxs and change: <option key="Current keyboard" value="false"/> <option key="Custom keyboard layout" value="empty"/> <option key="Grab keyboard" value="false"/> I have tried this on three NX Servers and all are fixed. Hope it helps or gets you closer. Always check in the ~/.nx/temp/ for the sshlog and see if --keyboard="empty/empty" instead of "pc105/en" because the Mac is really pc104. 9:05:35: startsession --session="HOST" --type="unix-gnome" --cache="8M" --images="32M" --link="adsl" --geometry="2556\ x1396" --screeninfo="2560x1440x32+render" --keyboard="empty/empty" --backingstore="1" --encryption="1" --composite="1" --\ shmem="1" --shpix="1" --streaming="1" --samba="0" --cups="0" --nodelay="1" --defer="0" --client="macosx" --media="0" --st\ rict="0" --aux="1"

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  • Webrick transparent proxy

    - by zzeroo
    Hi there, I've a absolute simple proxy running. require 'webrick' require 'webrick/httpproxy' s = WEBrick::HTTPProxyServer.new(:Port => 8080, :RequestCallback => Proc.new{|req,res| puts req.request_line, req.raw_header}) # Shutdown functionality trap("INT"){s.shutdown} # run the beast s.start This should in my mind not influence the communication in any way. But some sites doesn't work any more. Specially http://lastfm.de 's embedded flash players doesn't work. The header looks link: - -> http://ext.last.fm/2.0/?api%5Fsig=aa3e9ac9edf46ceb9a673cb76e61fef4&flashresponse=true&y=1269686332&streaming=true&playlistURL=lastfm%3A%2F%2Fplaylist%2Ftrack%2F42620245&fod=true&sk=ee93ae4f438767bf0183d26478610732&lang=de&api%5Fkey=da6ae1e99462ee22e81ac91ed39b43a4&method=playlist%2Efetch GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1 Host: play.last.fm User-Agent: Mozilla/5.0 (X11; U; Linux i686; de; rv:1.9.2) Gecko/20100308 Ubuntu/10.04 (lucid) Firefox/3.6 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: de,en-us;q=0.7,en;q=0.3 Accept-Encoding: gzip,deflate Accept-Charset: ISO-8859-1,utf-8;q=0.7,*;q=0.7 Keep-Alive: 115 Proxy-Connection: keep-alive Cookie: AnonWSSession=ee93ae4f438767bf0183d26478610732; AnonSession=cb8096e3b0d8ec9f4ffd6497a6d052d9-12bb36d49132e492bb309324d8a4100fc422b3be9c3add15ee90eae3190db5fc localhost - - [27/Mar/2010:11:38:52 CET] "GET http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout HTTP/1.1" 404 7593 - -> http://www.lastfm.de/log/flashclient/minor/Track_Loading_Fail/Buffering_Timeout localhost - - [27/Mar/2010:11:38:52 CET] "GET http://play.last.fm/preview/118270350.mp3 HTTP/1.1" 302 0 I nead some hints why or what the communication disturb.

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  • Executing bat file and returning the prompt

    - by Lieven Cardoen
    I have a problem with cruisecontrol where an ant scripts executes a bat file that doesn't give me the prompt back. As a result, the project in cruisecontrol keeps on bulding forever until I restart cruisecontrol. How can I resolve this? It's a startup.bat from wowza (Streaming Server) that I'm executing: @echo off call setenv.bat if not %WMSENVOK% == "true" goto end set _WINDOWNAME="Wowza Media Server 2" set _EXESERVER= if "%1"=="newwindow" ( set _EXESERVER=start %_WINDOWNAME% shift ) set CLASSPATH="%WMSAPP_HOME%\bin\wms-bootstrap.jar" rem cacls jmxremote.password /P username:R rem cacls jmxremote.access /P username:R rem NOTE: Here you can configure the JVM's built in JMX interface. rem See the "Server Management Console and Monitoring" chapter rem of the "User's Guide" for more information on how to configure the rem remote JMX interface in the [install-dir]/conf/Server.xml file. set JMXOPTIONS=-Dcom.sun.management.jmxremote=true rem set JMXOPTIONS=%JMXOPTIONS% -Djava.rmi.server.hostname=192.168.1.7 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.port=1099 rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.authenticate=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.ssl=false rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.password.file= "%WMSCONFIG_HOME%/conf/jmxremote.password" rem set JMXOPTIONS=%JMXOPTIONS% -Dcom.sun.management.jmxremote.access.file= "%WMSCONFIG_HOME%/conf/jmxremote.access" rem log interceptor com.wowza.wms.logging.LogNotify - see Javadocs for ILogNotify %_EXESERVER% "%_EXECJAVA%" %JAVA_OPTS% %JMXOPTIONS% -Dcom.wowza.wms.AppHome="%WMSAPP_HOME%" -Dcom.wowza.wms.ConfigURL="%WMSCONFIG_URL%" -Dcom.wowza.wms.ConfigHome="%WMSCONFIG_HOME%" -cp %CLASSPATH% com.wowza.wms.bootstrap.Bootstrap start :end

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  • Internet Explorer Warning when embedding Youtube on HTTPS site?

    - by pellepim
    Our application is run over HTTPS which rarely presents any problems for us. When it comes to youtube however, the fact that they do not present any content over SSL connections is giving us some head ache when trying to embed clips. Mostly because of Internet Explorers famous little warning message: "Do you want to view only the webpage content that was delivered securely? This page contains content that will not be delivered using a secure HTTPS ... etc" I've tried to solve this in several ways. The most promising one was to use the ProxyPass functionality in Apache to map to YouTube. Like this: ProxyPass: /youtube/ http://www.youtube.com ProxyPassReverse: /youtube/ http://www.youtube.com This gets rid of the annoying warning. However, the youtube SWF fails to start streaming The SWF i manage to load into the browser simply states : "An error occurred, please try again later". Potential solutions are perhaps: Download youtube FLV:s and serve them out of own domain (gah) Use custom FLV-player and stream only FLV:s from youtube over a https proxy? Update 10 March: I've tried to use Googles Youtube API for ActionScript to load a player. It looked promising at first and I was able to load a player through my https:// proxy. However, the SWF that is loaded contains loads of explicit calls to different non-ssl urls to create authentication links for the FLV-stream and for loading different crossdomain policies. It really seems like we're not supposed to access flv-streams directly. This makes it very hard to bypass the Internet Explorer warning, short of ripping out the FLV:s from youtube and serving them out of your own domain. There are solutions out there for downloading youtubes FLV:s. But that is not compliant with the Youtube terms of use and is really not an option for us.

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  • Should I move big data blobs in JSON or in separate binary connection?

    - by Amagrammer
    QUESTION: Is it better to send large data blobs in JSON for simplicity, or send them as binary data over a separate connection? If the former, can you offer tips on how to optimize the JSON to minimize size? If the latter, is it worth it to logically connect the JSON data to the binary data using an identifier that appears in both, e.g., as "data" : "< unique identifier " in the JSON and with the first bytes of the data blob being < unique identifier ? CONTEXT: My iPhone application needs to receive JSON data over the 3G network. This means that I need to think seriously about efficiency of data transfer, as well as the load on the CPU. Most of the data transfers will be relatively small packets of text data for which JSON is a natural format and for which there is no point in worrying much about efficiency. However, some of the most critical transfers will be big blobs of binary data -- definitely at least 100 kilobytes of data, and possibly closer to 1 megabyte as customers accumulate a longer history with the product. (Note: I will be caching what I can on the iPhone itself, but the data still has to be transferred at least once.) It is NOT streaming data. I will probably use a third-party JSON SDK -- the one I am using during development is here. Thanks

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  • Playing a .TS file on iOS

    - by Jonathan Grynspan
    We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

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  • Internet Radio Station for University

    - by ryan
    I am trying to help my University Student Radio station rethink the setup of the way they stream music, but I have some questions regarding the use of Ubuntu to stream music. Currently, the radio station uses two windows machines: one of which is used to stream the radio station and serve the website, and the other is used by rotating djs to select songs and create playlists. The computer used by djs feeds mono into the sound card of the server and the server streams the feed online. -Ideally I would like to maintain a two-computer setup: One computer as server, and another that is used to select and play music by rotating djs. -I would like to use Ubuntu for the server. -I would like to use Windows for the other machine. -The server should be able to stream song information. First, is there a way to somehow get the song information from an analog feed? Second, what is the best streaming server for radio? I have encountered shoutcast, icecast, and darwin, but I don't know where to begin in attempting to gauge them. Finally, if anyone has any tips or pointers about small internet radio station management/ setup they would be appreciated as this is my first radio station, and I am eager to hear of past experiences.

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  • Elegant Algorithm for Parsing Data Stream Into Record

    - by Matt Long
    I am interfacing with a hardware device that streams data to my app over Wifi. The data is streaming in just fine. The data contains a character header (DATA:) that indicates a new record has begun. The issues is that the data I receive doesn't necessarily fall on the header boundary, so I have to capture the data until what I've captured contains the header. Then, everything that precedes the header goes into the previous record and everything that comes after it goes into a new record. I have this working, but wondered if anyone has done this before and has a good computer-sciencey way to solve the problem. Here's what I do: Convert the NSData of the current read to an NSString Append the NSString to a placeholder string Check placeholder string for the header (DATA:). If the header is not there, just wait for the next read. If the header exists, append whatever precedes it to a previous record placeholder and hand that placeholder off to an array as a complete record that I can further parse into fields. Take whatever shows up after the header and place it in the record placeholder so that it can be appended to in the next read. Repeat steps 3 - 5. Let me know if you see any flaws with this or have a suggestion for a better way. Seems there should be some design pattern for this, but I can't think of one. Thanks.

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  • 4.0/WCF: Best approach for bi-idirectional message bus?

    - by TomTom
    Just a technology update, now that .NET 4.0 is out. I write an application that communicates to the server through what is basically a message bus (instead of method calls). This is based on the internal architecture of the application (which is multi threaded, passing the messages around). There are a limited number of messages to go from the client to the server, quite a lot more from the server to the client. Most of those can be handled via a separate specialized mechanism, but at the end we talk of possibly 10-100 small messages per second going from the server to the client. The client is supposed to operate under "internet conditions". THis means possibly home end users behind standard NAT devices (i.e. typical DSL routers) - a firewalled secure and thus "open" network can not be assumed. I want to have as little latency and as little overhad for the communication as possible. What is the technologally best way to handle the message bus callback? I Have no problem regularly calling to the server for message delivery if something needs to be sent... ...but what are my options to handle the messagtes from the server to the client? WsDualHttp does work how? Especially under a NAT scenario? Just as a note: polling is most likely out - the main problem here is that I would have a significant overhead OR a significant delay, both aren ot really wanted. Technically I would love some sort of streaming appraoch, where the server can write messags to a stream while he generates them and they get sent to the client as they come. Not esure this is doable with WCF, though (if not, I may acutally decide to handle the whole message part outside of WCF and just do control / login / setup / destruction via WCF).

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  • Rate limiting a ruby file stream

    - by Matthew Savage
    I am working on a project which involves uploading flash video files to a S3 bucket from a number of geographically distributed nodes. The video files are about 2-3mb each, and we are only sending one file (per node) every ten minutes, however the bandwidth we consume needs to be rate limited to ~20k/s, as these nodes are delivering streaming media to a CDN, and due to the locations we are only able to get 512k max upload. I have been looking into the ASW-S3 gem and while it doesn't offer any kind of rate limiting I am aware that you can pass in a IO Stream. Given this I am wondering if it might be possible to create a rate-limited stream which overrides the read method, adds in the rate limiting logic (e.g. in its simplest form a call to sleep between reads) and then call out to the super of the overridden method. Another option I considered is hacking the code for Net::HTTP and putting the rate limiting into the send_request_with_body_stream method which is using a while loop, but I'm not entirely sure which would be the best option. I have attempted at extending the IO class, however that didn't work at all, simply inheriting from the class with class ThrottledIO < IO didn't do anything. Any suggestions will be greatly appreciated.

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  • Database design for a media server containing movies, music, tv and everything in between?

    - by user364114
    In the near future I am attempting to design a media server as a personal project. MY first consideration to get the project underway is architecture, it will certainly be web based but more specifically than that I am looking for suggestions on the database design. So far I am considering something like the following, where I am using [] to represent a table, the first text is the table name to give an idea of purpose and the items within {} would be fields of the table. Also not, fid is functional id referencing some other table. [Item {id, value/name, description, link, type}] - this could be any entity, single song or whole music album, game, movie - almost see this as a recursive relation, ie. a song is an item but an album that song is part of is also an item or for example a tv season is an item, with multiple items being tv episodes [Type {id, fid, mime type, etc}] - file type specific information - could identify how code handles streaming/sending this item to a user [Location {id, fid, path to file?}] [Users {id, username, email, password, ...? }] - user account information [UAC {id, fid, acess level}] - i almost feel its more flexible to seperate access control permissions form the user accounts themselves [ItemLog {id, fid, fid2, timestamp}] - fid for user id, and fid2 for item id - this way we know what user access what when [UserLog {id, fid, timestamp}] -both are logs for access, whether login or last item access [Quota {id, fid, cap}] - some sort of way to throttle users from queing up the entire site and letting it download ... Suggestions or comments are welcome as the hope is that this project will be a open source project once some code is laid out.

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  • New TabItem Content ActualHeight crashes Xaml Window

    - by Jack Navarro
    I am able to create new TabItems with Content dynamically to a new window by streaming the Xaml with XamlReader: NewWindow newWindow = new NewWindow(); newWindow.Show(); TabControl myTabCntrol = newWindow.FindName("GBtabControl") as TabControl; StringReader stringReader = new StringReader(XamlGrid); XmlReader xmlReader = XmlReader.Create(stringReader); TabItem myTabItem = new TabItem(); myTabItem.Header = qDealName; myTabItem.Content = (UIElement)XamlReader.Load(xmlReader); myTabCntrol.Items.Add(myTabItem); This works fine. It displays a new grid wrapped in a scrollviewer. The problem is access the TabItem content from the newWindow. TabItem ti = GBtabControl.SelectedItem as TabItem; string scrollvwnm = "scrollViewer" + ti.Header.ToString(); MessageBox.Show(ti.ActualHeight.ToString()); // returns 21.5 ScrollViewer scrlvwr = this.FindName(scrollvwnm) as ScrollViewer; MessageBox.Show(scrollvwnm); // Displays name double checked for accuracy MessageBox.Show(scrlvwr.ActualHeight.ToString()); //Crashes ScrollViewer scrlvwr = ti.FindName(scrollvwnm) as ScrollViewer; MessageBox.Show(scrollvwnm); // Displays name double checked for accuracy MessageBox.Show(scrlvwr.ActualHeight.ToString()); //Also Crashes Is there a method to refresh UI in XAML so the new window is able to access the newly loaded tab item content? Thanks

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