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  • Progressbar for mediaplayer using jquery

    - by Geetha
    In my asp.net application i am using mediaplayer to play the audio and video. i am controling volume using javascript code. I want to display a userdefined progress bar. How to create control it. Code: <object id="mediaPlayer" classid="clsid:22D6F312-B0F6-11D0-94AB-0080C74C7E95" codebase="http://activex.microsoft.com/activex/controls/mplayer/en/nsmp2inf.cab#Version=5,1,52,701" height="1" standby="Loading Microsoft Windows Media Player components..." type="application/x-oleobject" width="1"> <param name="fileName" value="" /> <param name="animationatStart" value="true" /> <param name="transparentatStart" value="true" /> <param name="autoStart" value="true" /> <param name="showControls" value="true" /> <param name="volume" value="100" /> <param name="loop" value="true" /> </object>

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  • SIP and Java, where to start and with what?

    - by Senne
    I want to implement the SIP protocol in java and would want to be able to create different clients (5 or more) and make them connect to a proxy server. This is all for testing purposes so I would like to be able to see well what's happening on a rather low level. The clients should first be able to communicate trough text and later on maybe also by audio. (If I ever get that far) I already read a bit about the JAIN libraries and what I understood from that is that they are not really well suited for the server side? I also didn't really find any proxy server examples, tutorials, using JAIN. I also found this SIP Servlet Tutorial book, I used HTTP servlets in the past but should I prefer servlets or JAIN or ...? I'm quite new to SIP so I don't really know where to start or what to choose in combination with java.

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  • Java Media Framework always generating multicast packets with TTL=1

    - by Liam
    I need to generate a G711 multicast audio stream, and came across the AVTransmit2 sample as part of the Java Media Framework. Fundementally this works, however the multicast packets all have TTL set to 1. I found some documentation that suggested the SessionAddress could specify a TTL value, so I've tried changing that i.e. destAddr = new SessionAddress( ipAddr, port, 255); I also found some comments that the problems might be due to java defaulting to IPv6, so I've tried to force it to ipv4 by starting it like this: java -Djava.net.preferIPv4Stack=true -classpath "." AVTransmit2 javasound://8000 239.1.10.65 20480 However looking in wireshark, the packets still have TTL=1 I'm using JMF2.1.1e Any suggestions how to resolve this?

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  • Pure C# open source PCM to Ogg convertor?

    - by Ole Jak
    Microsoft Silverlight 4 is in beta. It supports PCM audio output. It would be madness to stream PCM over internet (for ex in P2P chart webApp) so we need Pure C# open source PCM to Ogg convertor. No unmanaged code, nothing going out of .net sandbox. So does any one know such Pure C# open source PCM to Ogg convertor? What do I need: Open Source Libs for encoding. Tutorials and blog articles on How to do it, about etc. BTW: why Pure C#? - because Silverlight 4 does not support unmanaged or just not C# DLL's. BTW2: this question is similar to this one but it is different because Ogg is Open Source, free while mp3 will not be free until 2010

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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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  • Javascript: Mediaplayer and its Progress Bar

    - by Geetha
    Hi All, In my asp.net application i am using mediaplayer to paly the audio and video. i am controling volume using javascript code. I want to display a userdefined progress bar. How to create it. Code: <object id="mediaPlayer" classid="clsid:22D6F312-B0F6-11D0-94AB-0080C74C7E95" codebase="http://activex.microsoft.com/activex/controls/mplayer/en/nsmp2inf.cab#Version=5,1,52,701" height="1" standby="Loading Microsoft Windows Media Player components..." type="application/x-oleobject" width="1"> <param name="fileName" value="" /> <param name="animationatStart" value="true" /> <param name="transparentatStart" value="true" /> <param name="autoStart" value="true" /> <param name="showControls" value="true" /> <param name="volume" value="100" /> <param name="loop" value="true" /> </object>

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  • Embed VLC player in GWT

    - by chrisnfoneur
    Hello, I want to embed a VLC player in my webapp build with Google's GWT. First I had a look at this page: http://wiki.videolan.org/GWT, which offers a nice solution but I add to implements all javascript functions calls (play, stop, fullscreen) with JSNI. Then I found gwt-player (hosted by Google code) which does all the job for me but the annoying part is that the project is not widely used (few posts each month on the project's group, not so many talks about it in blogs/forums...) Do you know another option to easly embed & control a VLC player in a GWT app ? My main goal is to play any video/audio file in a webapp and offer the user a fast/forward feature (set rate in VLC), is there any other player I could use ? I already had a look at Quicktime, Windows Media player & Flowplayer, none of them offers as much features as VLC. Thanks in advance & have a nice new year's eve. Chris

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  • No Microphone error on iPod Touch

    - by Bob Vork
    I've build an iPhone app that should work on an iPod Touch as well, but I'm getting reports that the app is not working on iPod touches. It's displaying an error message saying there's no mic available on the device. The thing is, the app does nothing whatsoever with audio, and I can't find anything related in the project settings. The other problem is I don't have an iPod Touch available to test this myself. Are some people running an old firmware version? Am I compiling the wrong firmware version? To my surprise I couldn't find anything about this on SO or Google… Any help is appreciated

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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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  • Is there a DRM scheme that works?

    - by Simon
    We help our clients to manage and publish their media online - images, video, audio, whatever. They always ask my boss whether they can stop users from copying their media, and he asks me, and I always tell him the same thing: no. If the users can view the media, then a sufficiently determined user will always be able to make a copy. But am I right? I've been asked again today, and I promised my boss I'd ask about it online. So - is there a DRM scheme that will work? One that will stop users making copies without stopping legitimate viewing of the media? And if there isn't, how do I convince my boss?

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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 to create playable FLV video from part of FLV file using FFMPEG?

    - by Ole Jak
    So we had real FLV video file. we had devided it into 3 parts (more or less equal, not looking into structure orcontext). We have taken second part and forgot about first 2. Video contained audio and video track. mp3 and on vp6. Is it any how possible to play thsat second part after sending to ffmpeg some command? So how to (using any FFMPEG API (in general in any programming language) or using command line) turn bytearray into playable video? (knowing what format video was created in and some other data like used codecs )

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  • win7 amd64 guest in kvm does not have sound

    - by davidshen84
    hi, my host system is gentoo amd64, guest system is win 7 amd64. the guest system can work, except it does not have sound. i start kvm with -soundhw ac97, QEMU_AUDIO_DRV='alsa', and after i get into the guest system, i can see a 'Multimedia Audio Controller' in the device manager. but win7 cannot find the driver for it. i searched the network for a long time, and i cannot find a driver for intel ac97 for win7 amd64. i also tried -soundhw sb16, es1370, none of them work. please help me fix this.

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  • fourier transform to transpose key of a wav file

    - by tbischel
    I want to write an app to transpose the key a wav file plays in (for fun, I know there are apps that already do this)... my main understanding of how this might be accomplished is to 1) chop the audio file into very small blocks (say 1/10 a second) 2) run an FFT on each block 3) phase shift the frequency space up or down depending on what key I want 4) use an inverse FFT to return each block to the time domain 5) glue all the blocks together But now I'm wondering if the transformed blocks would no longer be continuous when I try to glue them back together. Are there ideas how I should do this to guarantee continuity, or am I just worrying about nothing?

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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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  • How to make QT support HTML 5 database?

    - by Mickey Shine
    I am using Qt 4.7.1 and embedded a webview in my app. But I got the following error when trying to visit http://webkit.org/demos/sticky-notes/ to test the HTML 5 database feature Failed to open the database on disk. This is probably because the version was bad or there is not enough space left in this domain's quota I compiled my static Qt library with the following command: configure --prefix=/usr/local/qt-static-release-db --accessibility --multimedia --audio-backend --svg --webkit --javascript-jit --script --scripttools --declarative --release -nomake examples -nomake demos --static --openssl -I /usr/local/ssl/include -L /usr/local/ssl/lib -confirm-license -sql-qsqlite -sql-qmysql -sql-qodbc

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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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  • How do I force one method to be executed before another method?

    - by RexOnRoids
    I've got 2 methods. One method starts playing an audio file (.mp3), the other method updates a UIToolBar to show a button (PLAY or PAUSE). These two methods are called in the following order: //Adds some UIBarButtonItems to a UIToolBar [self togglePlayer]; //Uses AVAudioPlayer [audioPlayer play]; I call the methods in the above order so that the (pause) button will be shown at the time the song starts playing. But, the problem is that the song starts playing first, and the UIToolBar remains unchanged for quite a while (from 2 to 5 secs) until the button is added and shown. What I want is for the button to be shown at the same time the song starts playing (i.e. NO DELAY). Is there any way to do this?

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  • ActionController::MethodNotAllowed

    - by Lowgain
    I have a rails model called 'audioclip'. I orginally created a scaffold with a 'new' action, which I replaced with 'new_record' and 'new_upload', becasue there are two ways to attach audio to this model. Going to /audioclips/new_record doesn't work because it takes 'new_record' as if it was an id. Instead of changing this, I was trying to just create '/record_clip' and '/upload_clip' paths. So in my routes.db I have: map.record_clip '/record_clip', :controller => 'audioclips', :action => 'new_record' map.upload_clip '/upload_clip', :controller => 'audioclips', :action => 'new_upload' When I navigate to /record_clip, I get ActionController::MethodNotAllowed Only get, head, post, put, and delete requests are allowed. I'm not extremely familiar with the inner-workings of routing yet. What is the problem here? (If it helps, I have these two statements above map.resources = :audioclips

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  • DirectSound affects system volume on WinXP

    - by Anton
    Hello guys, I'm currently developing an audio engine that is used in voice network chat software. Everything is working fine - capture/playback/mixing channels. The problem is in using it under Windows XP. I've been getting user reports with information that their global system volume is set to zero after launching the application. I'm assuming that happens because of WaveOut/DSound conflict. How can I force DSound not to affect system volume? Playback device is initialized: DirectSoundCreate8(&GUID, &pAudio, NULL); and: pAudio-SetCooperativeLevel(parentWnd, DSSCL_PRIORITY); I'm currently not able to debug the application, cause I'm using Vista and everything is OK. Hope you can help me with this issue! Thanks a Lot! Regards, Anton.

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  • Alternative to Rtmp and red5 for Iphone application

    - by IeN
    I am using red5 + rtmp in client-server flash application. There isnt audio/video streams in my applications, rtmp used for transfering messages from app to server and back. Now i need to develop application for Iphone and need help: 1) is there any rtmp implementation on Iphone ?? 2) If not, how could i solve this problem? Is there is any alternative to rtmp on iphone? And most important question : could it be solved without rewriting whole server part of application? (red5+ rtmp) Thanks

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  • Fast block placement algorithm, advice needed?

    - by James Morris
    I need to emulate the window placement strategy of the Fluxbox window manager. As a rough guide, visualize randomly sized windows filling up the screen one at a time, where the rough size of each results in an average of 80 windows on screen without any window overlapping another. It is important to note that windows will close and the space that closed windows previously occupied becomes available once more for the placement of new windows. The window placement strategy has three binary options: Windows build horizontal rows or vertical columns (potentially) Windows are placed from left to right or right to left Windows are placed from top to bottom or bottom to top Why is the algorithm a problem? It needs to operate to the deadlines of a real time thread in an audio application. At this moment I am only concerned with getting a fast algorithm, don't concern yourself over the implications of real time threads and all the hurdles in programming that that brings. So far I have two choices which I have built loose prototypes for: 1) A port of the Fluxbox placement algorithm into my code. The problem with this is, the client (my program) gets kicked out of the audio server (JACK) when I try placing the worst case scenario of 256 blocks using the algorithm. This algorithm performs over 14000 full (linear) scans of the list of blocks already placed when placing the 256th window. 2) My alternative approach. Only partially implemented, this approach uses a data structure for each area of rectangular free unused space (the list of windows can be entirely separate, and is not required for testing of this algorithm). The data structure acts as a node in a doubly linked list (with sorted insertion), as well as containing the coordinates of the top-left corner, and the width and height. Furthermore, each block data structure also contains four links which connect to each immediately adjacent (touching) block on each of the four sides. IMPORTANT RULE: Each block may only touch with one block per side. The problem with this approach is, it's very complex. I have implemented the straightforward cases where 1) space is removed from one corner of a block, 2) splitting neighbouring blocks so that the IMPORTANT RULE is adhered to. The less straightforward case, where the space to be removed can only be found within a column or row of boxes, is only partially implemented - if one of the blocks to be removed is an exact fit for width (ie column) or height (ie row) then problems occur. And don't even mention the fact this only checks columns one box wide, and rows one box tall. I've implemented this algorithm in C - the language I am using for this project (I've not used C++ for a few years and am uncomfortable using it after having focused all my attention to C development, it's a hobby). The implementation is 700+ lines of code (including plenty of blank lines, brace lines, comments etc). The implementation only works for the horizontal-rows + left-right + top-bottom placement strategy. So I've either got to add some way of making this +700 lines of code work for the other 7 placement strategy options, or I'm going to have to duplicate those +700 lines of code for the other seven options. Neither of these is attractive, the first, because the existing code is complex enough, the second, because of bloat. The algorithm is not even at a stage where I can use it in the real time worst case scenario, because of missing functionality, so I still don't know if it actually performs better or worse than the first approach. What else is there? I've skimmed over and discounted: Bin Packing algorithms: their emphasis on optimal fit does not match the requirements of this algorithm. Recursive Bisection Placement algorithms: sounds promising, but these are for circuit design. Their emphasis is optimal wire length. Both of these, especially the latter, all elements to be placed/packs are known before the algorithm begins. I need an algorithm which works accumulatively with what it is given to do when it is told to do it. What are your thoughts on this? How would you approach it? What other algorithms should I look at? Or even what concepts should I research seeing as I've not studied computer science/software engineering? Please ask questions in comments if further information is needed. [edit] If it makes any difference, the units for the coordinates will not be pixels. The units are unimportant, but the grid where windows/blocks/whatever can be placed will be 127 x 127 units.

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  • Testing MPMoviePlayerViewController in iPad simulator

    - by hgpc
    I have a view that shows a MPMoviePlayerViewController modally. When testing it in the iPad simulator it works well on the first try. If I dismiss the video and then show the view again, the player only plays the audio, but not the video. Is this a simulator quirk or am I doing something wrong? Here's my code: - (void)viewWillAppear:(BOOL)animated { [super viewWillAppear:animated]; MPMoviePlayerViewController* v = [[MPMoviePlayerViewController alloc] initWithContentURL:url]; [[NSNotificationCenter defaultCenter] addObserver:self selector: @selector(playbackDidFinish:) name:MPMoviePlayerPlaybackDidFinishNotification object:v.moviePlayer]; [self presentMoviePlayerViewControllerAnimated:v]; [v release]; } -(void) playbackDidFinish:(NSNotification*)aNotification { MPMoviePlayerController *player = [aNotification object]; [[NSNotificationCenter defaultCenter] removeObserver:self name:MPMoviePlayerPlaybackDidFinishNotification object:player]; [player stop]; [self dismissMoviePlayerViewControllerAnimated]; }

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