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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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • 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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  • super dealloc error using multiple table view calsses

    - by padatronic
    I am new to Iphone apps and I am trying to build a tab based app. I am attempting to have a table ontop of an image in both tabs. On tab with a table of audio links and the other tab with a table of video links. This has all gone swimmingly, I have created two viewControllers for the two tables. All the code works great apart from to get it to work I have to comment out the super dealloc in the - (void)dealloc {} in the videoTableViewController for the second tab. If I don't I get the error message: FREED(id): message numberOfSectionsInTableView: sent to freed object please help, i have no idea why it is doing this...

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  • AVAudioPlayer Output to Speaker Problem

    - by Max
    After searching around for how to send AVAudioPlayer output to the iPhone's speaker, I found this: http://stackoverflow.com/questions/1064846/iphone-audio-playback-force-through-internal-speaker Despite setting the category correctly to AVAudioSessionCategoryPlayAndRecord, this solution doesn't seem to be working for me and won't even let the build compile, giving me this error: "_AudioSessionSetProperty", referenced from: ... ... ld: symbol(s) not found collect2: ld returned 1 exit status Am I not including something? I'm importing AudioToolbox, AVFoundation, and CoreAudio. My class implements AVAudioSessionDelegate, AVAudioRecorderDelegate, AVAudioPlayerDelegate, and UITextFieldDelegate. Any help would be greatly appreciated!

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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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