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  • Podcast Best Practices - Page Development & Monetization Considerations

    - by Christopher Ickes
    Our current podcast page has show notes and a link to download an mp3 of our podcast. We were advised to add an audio player to stream the file live from our website. The thought being this would improve time spent on our site and allow for greater advertising dollars. Is it better to have a page with show notes, an mp3 for download AND also stream the podcast live OR just stick to the show notes & mp3 download? Does anyone see any affect on advertising revenue, either way?

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  • Mplayer can't play *.wmv file

    - by Jimmy
    I have a problem when I use the mplayer to play *.wmv file on my ubuntu11.10. There are some error messages here. Could anyone can help me solve this problem. I use some keyword to search in Gooele, but I can't find the answer. Thank you. Playing testmovie.wmv. ASF file format detected. [asfheader] Audio stream found, -aid 1 [asfheader] Video stream found, -vid 2 VIDEO: [WMV3] 1280x720 24bpp 1000.000 fps 4000.0 kbps (488.3 kbyte/s) Load subtitles in ./ open: No such file or directory [ MGA] Couldn't open: /dev/mga_vid open: No such file or directory [MGA] Couldn't open: /dev/mga_vid [VO_TDFXFB] Can't open /dev/fb0: Permission denied. [VO_3DFX] Unable to open /dev/3dfx. [vdpau] Error when calling vdp_device_create_x11: 1 ========================================================================== Opening video decoder: [dmo] DMO video codecs DMO dll supports VO Optimizations 0 1 DMO dll might use previous sample when requested MPlayer interrupted by signal 11 in module: init_video_codec I am using xv as my video driver.

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  • Buddypress with bbpress: Showing latest topics on front page

    - by MadsMadsDk
    I'm doing a wordpress solution with BuddyPress and bbPress, where I need to display the five newest topics on the homepage, as if it was blog-entries, but it seems kind of hard to accomplish. I'm figuring I gotta do something with the activity stream, but it seems like the stream is based on the user who is currently logged in, which is not what I want. So what should I do? Use a nifty plugin that does the trick (maybe someone knows a plugin I don't know of, as I've already tried the bbPress Latest Discussion plugin) Hardcode a forum-activity loop into the page-template file, using the is_front_page() function? Is there a forum-activity hook, that display the latest forum topics sitewide? Thanks in advance

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  • How To Specify Bitrate, Codec and Demultiplexing for VLC Video Capture or Recording

    - by Subhash
    I capture video from old TV tuner card - Pinnacle PCTV - using VLC. The video is from the Composite input and audio is from I guess the mixer or Line in. The command I use is: vlc v4l2:///dev/video0:normal=pal:width=720:height=576:input=1 :input-slave="alsa://hw:0,0" In VLC, I have enabled the Advanced Controls toolbar, which allows me to record videos when I want to. However, these videos are uncompressed - very big and play only with VLC. Totem throws the "Could not demultiplex stream" error. I need to convert them using WinFF to reduce their size and make them playable with Totem and other software. My question is whether I can configure the recording settings - the codecs and the bitrate, and also get the stream demultiplexed. If I pass any -sout parameter with command I get a "Segmentation fault". I use 64-bit Ubuntu 10.10.

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  • Youtube API upload - Incomplete Multipart body error

    - by Blerim J
    Hello, I'm trying to upload videos in Youtube through HttpWebRequest. Everything seems to be fine when uploading following the example given in API documentation. I see that request is being formed correctly, with content and token sent but I receive "Incomplete multipart body" as response. Thanks Blerim public bool YouTubeUpload() { string newLine = "\r\n"; //token and url are retrieved from YouTube at runtime. string token = string.Empty; string url = string.Empty; // construct the command url url = url + "?nexturl=http://www.mywebsite.com/"; // get a unique string to use for the data boundary string boundary = Guid.NewGuid().ToString().Replace("-", string.Empty); foreach (string file in Request.Files) { HttpPostedFileBase hpf = Request.Files[file] as HttpPostedFileBase; if (hpf.ContentLength == 0) continue; // get info about the file and open it for reading Stream fs = hpf.InputStream; HttpWebRequest webRequest = (HttpWebRequest)WebRequest.Create(url); webRequest.ContentType = "multipart/form-data; boundary=" + boundary; webRequest.Method = "POST"; webRequest.KeepAlive = true; webRequest.Credentials = System.Net.CredentialCache.DefaultCredentials; MemoryStream memoryStream = new MemoryStream(); StreamWriter writer = new StreamWriter(memoryStream); //token writer.Write("--" + boundary + newLine); writer.Write("Content-Disposition: form-data; name=\"{0}\"{1}{2}", "token", newLine, newLine); writer.Write(token); writer.Write(newLine); //Video writer.Write("--" + boundary + newLine); writer.Write("Content-Disposition: form-data; name=\"{0}\"; filename=\"{1}\"{2}", "File1", hpf.FileName, newLine); writer.Write("Content-Type: {0}" + newLine + newLine, hpf.ContentType); writer.Flush(); byte[] boundarybytes = System.Text.Encoding.ASCII.GetBytes(string.Format("--{0}--{1}", boundary, newLine)); webRequest.ContentLength = memoryStream.Length + fs.Length + boundarybytes.Length; Stream webStream = webRequest.GetRequestStream(); // write the form data to the web stream memoryStream.Position = 0; byte[] tempBuffer = new byte[memoryStream.Length]; memoryStream.Read(tempBuffer, 0, tempBuffer.Length); memoryStream.Close(); webStream.Write(tempBuffer, 0, tempBuffer.Length); // write the file to the stream int size; byte[] buf = new byte[1024 * 10]; do { size = fs.Read(buf, 0, buf.Length); if (size > 0) webStream.Write(buf, 0, size); } while (size > 0); // write the trailer to the stream webStream.Write(boundarybytes, 0, boundarybytes.Length); webStream.Close(); fs.Close(); //fails here. Error - Incomplete multipart body. WebResponse webResponse = webRequest.GetResponse(); } return true; }

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  • How to download streaming video from a flash player (arte)?

    - by wim
    there are a couple of concerts that I would very much like to view from Arte (I think it's a french TV channel?) but my connection is not good enough to stream the video. How can I download the file to play it locally? Here is an example link: http://www.my-jazzlive.tv/?p=1862 I have tried popular browser plugins such as DownloadHelper and Flash Video Downloader, these are working fine for me on sites such as youtube but they don't seem to recognise any stream from the Arte video player. I also looked through /tmp for something that looks like a partially downloaded video, but no luck.

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  • Is SOAP Http POST more complicated than I thought

    - by Pete Petersen
    I'm currently writing a bit of code to send some xml data to a web service via HTTP POST. I thought this would be really simple and have written the following example code (C#) Console.WriteLine("Press enter to send data..."); while (Console.ReadLine() != "q") { HttpWebRequest httpWReq = (HttpWebRequest)WebRequest.Create(@"http://localhost:8888/"); Foo fooItem = new Foo { Member1 = "05", Member2 = "74455604", Member3 = "15101051", Member4 = 1, Member5 = "fsf", Member6 = 6.52, }; ASCIIEncoding encoding = new ASCIIEncoding(); string postData = fooItem.ToXml(); byte[] data = encoding.GetBytes(postData); httpWReq.Method = "POST"; httpWReq.ContentType = "application/xml"; httpWReq.ContentLength = data.Length; using (Stream stream = httpWReq.GetRequestStream()) { stream.Write(data, 0, data.Length); } HttpWebResponse response = (HttpWebResponse)httpWReq.GetResponse(); string responseString = new StreamReader(response.GetResponseStream()).ReadToEnd(); Console.WriteLine("Received " + responseString); Console.WriteLine("Press enter to send data..."); } This is all I thought would be necessary, however I have now been given the details for the web service. This included some information which is unfarmiliar to me and I'm unsure whether I need to include it. The information I was sent was <url>http://sometext/soap/rpc</url> <namespace>http://sometext/a.services</namespace> <method>receiveInfo</method> <parm-id>xmldata</parm-id> (Input data) (Actual XML data as string) <parm-id>status</parm-id> (Output data) <userid>user</userid> <password>pass</password> <secure>false</secure> I guess this means I need to include a username and password somehow, but I'm not sure what the namespace or method fields are used for. Could anyone give me a hint? Sorry I've never used webservices before.

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  • Cannot decode complete cipher list in .NET SslStream handshake.

    - by karmasponge
    While attempting to move from a 'C' based SSL implementation to C# using the .NET SslStream and we have run into what look like cipher compatibility issues with the .NET SslStream and a AS400 machine we are trying to connect to (which worked previously). When we call SslStream.AuthenticateAsClient it is sending the following: 16 03 00 00 37 01 00 00 33 03 00 4d 2c 00 ee 99 4e 0c 5d 83 14 77 78 5c 0f d3 8f 8b d5 e6 b8 cd 61 0f 29 08 ab 75 03 f7 fa 7d 70 00 00 0c 00 05 00 0a 00 13 00 04 00 02 00 ff 01 00 Which decodes as (based on http://www.mozilla.org/projects/security/pki/nss/ssl/draft302.txt) [16] Record Type [03 00] SSL Version [00 37] Body length [01] SSL3_MT_CLIENT_HELLO [00 00 33] Length (51 bytes) [03 00] Version number = 768 [4d 2c 00 ee] 4 Bytes unix time [… ] 28 Bytes random number [00] Session number [00 0c] 12 bytes (2 * 6 Cyphers)? [00 05, 00 0a, 00 13, 00 04, 00 02, 00 ff] - [RC4, PBE-MD5-DES, RSA, MD5, PKCS, ???] [01 00] Null compression method The as400 server responds back with: 15 03 00 00 02 02 28 [15] SSL3_RT_ALERT [03 00] SSL Version [00 02] Body Length (2 Bytes) [02 28] 2 = SSL3_RT_FATAL, 40 = SSL3_AD_HANDSHAKE_FAILURE I'm specifically looking to decode the '00 FF' at the end of the cyphers. Have I decoded it correctly? What does, if anything, '00 FF' decode too? I am using the following code to test/reproduce: using System; using System.Collections.Generic; using System.Linq; using System.Text; using System.Net.Sockets; using System.Net.Security; using System.Security.Authentication; using System.IO; using System.Diagnostics; using System.Security.Cryptography.X509Certificates; namespace TestSslStreamApp { class DebugStream : Stream { private Stream AggregatedStream { get; set; } public DebugStream(Stream stream) { AggregatedStream = stream; } public override bool CanRead { get { return AggregatedStream.CanRead; } } public override bool CanSeek { get { return AggregatedStream.CanSeek; } } public override bool CanWrite { get { return AggregatedStream.CanWrite; } } public override void Flush() { AggregatedStream.Flush(); } public override long Length { get { return AggregatedStream.Length; } } public override long Position { get { return AggregatedStream.Position; } set { AggregatedStream.Position = value; } } public override int Read(byte[] buffer, int offset, int count) { int bytesRead = AggregatedStream.Read(buffer, offset, count); return bytesRead; } public override long Seek(long offset, SeekOrigin origin) { return AggregatedStream.Seek(offset, origin); } public override void SetLength(long value) { AggregatedStream.SetLength(value); } public override void Write(byte[] buffer, int offset, int count) { AggregatedStream.Write(buffer, offset, count); } } class Program { static void Main(string[] args) { const string HostName = "as400"; TcpClient tcpClient = new TcpClient(HostName, 992); SslStream sslStream = new SslStream(new DebugStream(tcpClient.GetStream()), false, null, null, EncryptionPolicy.AllowNoEncryption); sslStream.AuthenticateAsClient(HostName, null, SslProtocols.Ssl3, false); } } }

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  • Converting mp4 to mp3

    - by aki
    I have a video I need to convert to mp3 (from the command line - not GUI) video.mp4 I tried: ffmpeg -i -b 192 video.mp4 video.mp3 with no success. I get the following error: WARNING: library configuration mismatch Seems stream 0 codec frame rate differs from container frame rate: 59.83 (29917/500) -> 59.75 (239/4) WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s Encoder (codec id 86017) not found for output stream #0.0 so I tried lame: lame -h -b 192 video.mp4 video.mp3 I get: Warning: unsupported audio format Am I missing something?

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  • How to avoid or minimise use of check/conditional statement in my scenario?

    - by Muneeb Nasir
    I have scenario, where I got stream and I need to check for some value. If I got any new value I have to store it in any of data structure. It seems very easy, I can place conditional statement if-else or can use contain method of set/map to check either received is new or not. But the problem is checking will effect my application performance, in stream I will receive hundreds for value in second, if I start checking each and every value I received then for sure it effect performance. Anybody can suggest me any mechanism or algorithm to solve my issue, either by bypassing checks or at least minimize them?

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  • How do i use latest Pulseaudio in 11.10?

    - by YumYumYum
    Ubuntu 11.04 i had pulseaudio from source compiled and i used it to learn, it always worked (git versions). But since i have Ubuntu 11.10, i can install it but i can not use it anymore like i do in 11.04 before. Everytime i play something its throwing this: $ speaker-test speaker-test 1.0.24.2 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 192 to 2097152 Period size range from 64 to 699051 Using max buffer size 2097152 Periods = 4 ALSA lib pcm_pulse.c:746:(pulse_prepare) PulseAudio: Unable to create stream: Invalid argument Unable to set hw params for playback: Input/output error Setting of hwparams failed: Input/output error How to make pulseaudio work in 11.10 from source?

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  • Resumable upload from Java client to Grails web application?

    - by dersteps
    After almost 2 workdays of Googling and trying several different possibilities I found throughout the web, I'm asking this question here, hoping that I might finally get an answer. First of all, here's what I want to do: I'm developing a client and a server application with the purpose of exchanging a lot of large files between multiple clients on a single server. The client is developed in pure Java (JDK 1.6), while the web application is done in Grails (2.0.0). As the purpose of the client is to allow users to exchange a lot of large files (usually about 2GB each), I have to implement it in a way, so that the uploads are resumable, i.e. the users are able to stop and resume uploads at any time. Here's what I did so far: I actually managed to do what I wanted to do and stream large files to the server while still being able to pause and resume uploads using raw sockets. I would send a regular request to the server (using Apache's HttpClient library) to get the server to send me a port that was free for me to use, then open a ServerSocket on the server and connect to that particular socket from the client. Here's the problem with that: Actually, there are at least two problems with that: I open those ports myself, so I have to manage open and used ports myself. This is quite error-prone. I actually circumvent Grails' ability to manage a huge amount of (concurrent) connections. Finally, here's what I'm supposed to do now and the problem: As the problems I mentioned above are unacceptable, I am now supposed to use Java's URLConnection/HttpURLConnection classes, while still sticking to Grails. Connecting to the server and sending simple requests is no problem at all, everything worked fine. The problems started when I tried to use the streams (the connection's OutputStream in the client and the request's InputStream in the server). Opening the client's OutputStream and writing data to it is as easy as it gets. But reading from the request's InputStream seems impossible to me, as that stream is always empty, as it seems. Example Code Here's an example of the server side (Groovy controller): def test() { InputStream inStream = request.inputStream if(inStream != null) { int read = 0; byte[] buffer = new byte[4096]; long total = 0; println "Start reading" while((read = inStream.read(buffer)) != -1) { println "Read " + read + " bytes from input stream buffer" //<-- this is NEVER called } println "Reading finished" println "Read a total of " + total + " bytes" // <-- 'total' will always be 0 (zero) } else { println "Input Stream is null" // <-- This is NEVER called } } This is what I did on the client side (Java class): public void connect() { final URL url = new URL("myserveraddress"); final byte[] message = "someMessage".getBytes(); // Any byte[] - will be a file one day HttpURLConnection connection = url.openConnection(); connection.setRequestMethod("GET"); // other methods - same result // Write message DataOutputStream out = new DataOutputStream(connection.getOutputStream()); out.writeBytes(message); out.flush(); out.close(); // Actually connect connection.connect(); // is this placed correctly? // Get response BufferedReader in = new BufferedReader(new InputStreamReader(connection.getInputStream())); String line = null; while((line = in.readLine()) != null) { System.out.println(line); // Prints the whole server response as expected } in.close(); } As I mentioned, the problem is that request.inputStream always yields an empty InputStream, so I am never able to read anything from it (of course). But as that is exactly what I'm trying to do (so I can stream the file to be uploaded to the server, read from the InputStream and save it to a file), this is rather disappointing. I tried different HTTP methods, different data payloads, and also rearranged the code over and over again, but did not seem to be able to solve the problem. What I hope to find I hope to find a solution to my problem, of course. Anything is highly appreciated: hints, code snippets, library suggestions and so on. Maybe I'm even having it all wrong and need to go in a totally different direction. So, how can I implement resumable file uploads for rather large (binary) files from a Java client to a Grails web application without manually opening ports on the server side?

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  • Data architecture for event log metrics?

    - by elliot42
    My service has a large ongoing number of user events, and we would like to do things like "count occurrence of event type T since date D." We are trying to make two basic decisions: What to store? Storing every event vs. only storing aggregates (Event log style) log every event and count them later, vs. (Time-series style) store a single aggregated "count of event E for date D" for every day Where to store the data In a relational database (particularly MySQL) In a non-relational (NoSQL) database In flat log files (collected centrally over the network via syslog-ng) What is standard practice / where can I read more about comparing the different types of systems? Additional details: The total event stream is large, potentially hundreds of thousands of entries per day But our current need is only to count certain types of events within it We don't necessarily need real-time access to the raw data or aggregation results IMHO, "log all events to files, crawl them at a later time to filter and aggregate the stream" is a pretty standard UNIX Way, but my Rails-y compatriots seem to think that nothing is real unless it's in MySQL.

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  • how to avoid or minimise use of check/conditional statement?

    - by Muneeb Nasir
    I have scenario, where i got stream and i need to check for some value. if i got my any new value i have to store it in any of data structure. well it seems very easy, i can place conditional statement if-else or can use contain method of set/map to check either received is new or not. but the problem is checking will effect my application performance, in stream i'll receive hundreds for value in second, if i start checking each and every value i received than for sure it effect performance. Any body can suggest me any mechanism or algorithm that solve my issue. either by bypassing checks or atleast minimize them?

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  • Structuring an input file

    - by Ricardo
    I am in the process of structuring a small program to perform some hydraulic analysis of pipe flow. As I am envisioning this, the program will read an input file, store the input parameters in a suitable way, operate on them and finally output results. I am struggling with how to structure the input file in a sane way; that is, in a way that a human can write it easily and a machine can parse it easily. A sample input file made available to me for a similar program is just a stream of comma-separated numbers that don't make much sense on their own, so that's the scenario I am trying to avoid. Though I am giving the details of my particular problem, I am more interested in general input-file structuring strategies. Is a stream of comma-separated values my best bet? Would I be better off using some sort of key:value structure? I don't know much about this, so any help will probably put me in a better track than I am now.

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  • Bluetooth connection. Problem with sony ericsson.

    - by Hugi
    I have bt client and server. Then i use method Connector.open, client connects to the port, but passed so that my server does not see them. Nokia for all normal, but with sony ericsson i have this problem. On bt adapter open one port (com 5). Listings Client /* * To change this template, choose Tools | Templates * and open the template in the editor. */ import java.util.Vector; import javax.bluetooth.*; import javax.microedition.midlet.*; import javax.microedition.lcdui.*; import javax.microedition.io.*; import java.io.*; /** * @author ????????????? */ public class Client extends MIDlet implements DiscoveryListener, CommandListener { private static Object lock=new Object(); private static Vector vecDevices=new Vector(); private ServiceRecord[] servRec = new ServiceRecord[INQUIRY_COMPLETED]; private Form form = new Form( "Search" ); private List voteList = new List( "Vote list", List.IMPLICIT ); private List vote = new List( "", List.EXCLUSIVE ); private RemoteDevice remoteDevice; private String connectionURL = null; protected int stopToken = 255; private Command select = null; public void startApp() { //view form Display.getDisplay(this).setCurrent(form); try { //device search print("Starting device inquiry..."); getAgent().startInquiry(DiscoveryAgent.GIAC, this); try { synchronized(lock){ lock.wait(); } }catch (InterruptedException e) { e.printStackTrace(); } //device count int deviceCount=vecDevices.size(); if(deviceCount <= 0) { print("No Devices Found ."); } else{ remoteDevice=(RemoteDevice)vecDevices.elementAt(0); print( "Server found" ); //create uuid UUID uuid = new UUID(0x1101); UUID uuids[] = new UUID[] { uuid }; //search service print( "Searching for service..." ); getAgent().searchServices(null,uuids,remoteDevice,this); } } catch( Exception e) { e.printStackTrace(); } } //if deivce discovered add to vecDevices public void deviceDiscovered(RemoteDevice btDevice, DeviceClass cod) { //add the device to the vector try { if(!vecDevices.contains(btDevice) && btDevice.getFriendlyName(true).equals("serverHugi")){ vecDevices.addElement(btDevice); } } catch( IOException e ) { } } public synchronized void servicesDiscovered(int transID, ServiceRecord[] servRecord) { //for each service create connection if( servRecord!=null && servRecord.length>0 ){ print( "Service found" ); connectionURL = servRecord[0].getConnectionURL(ServiceRecord.NOAUTHENTICATE_NOENCRYPT,false); //connectionURL = servRecord[0].getConnectionURL(ServiceRecord.AUTHENTICATE_NOENCRYPT,false); } if ( connectionURL != null ) { showVoteList(); } } public void serviceSearchCompleted(int transID, int respCode) { //print( "serviceSearchCompleted" ); synchronized(lock){ lock.notify(); } } //This callback method will be called when the device discovery is completed. public void inquiryCompleted(int discType) { synchronized(lock){ lock.notify(); } switch (discType) { case DiscoveryListener.INQUIRY_COMPLETED : print("INQUIRY_COMPLETED"); break; case DiscoveryListener.INQUIRY_TERMINATED : print("INQUIRY_TERMINATED"); break; case DiscoveryListener.INQUIRY_ERROR : print("INQUIRY_ERROR"); break; default : print("Unknown Response Code"); break; } } //add message at form public void print( String msg ) { form.append( msg ); form.append( "\n\n" ); } public void pauseApp() { } public void destroyApp(boolean unconditional) { } //get agent :))) private DiscoveryAgent getAgent() { try { return LocalDevice.getLocalDevice().getDiscoveryAgent(); } catch (BluetoothStateException e) { throw new Error(e.getMessage()); } } private synchronized String getMessage( final String send ) { StreamConnection stream = null; DataInputStream in = null; DataOutputStream out = null; String r = null; try { //open connection stream = (StreamConnection) Connector.open(connectionURL); in = stream.openDataInputStream(); out = stream.openDataOutputStream(); out.writeUTF( send ); out.flush(); r = in.readUTF(); print( r ); in.close(); out.close(); stream.close(); return r; } catch (IOException e) { } finally { if (stream != null) { try { stream.close(); } catch (IOException e) { } } return r; } } private synchronized void showVoteList() { String votes = getMessage( "c_getVotes" ); voteList.append( votes, null ); select = new Command( "Select", Command.OK, 4 ); voteList.addCommand( select ); voteList.setCommandListener( this ); Display.getDisplay(this).setCurrent(voteList); } private synchronized void showVote( int index ) { String title = getMessage( "c_getVote_"+index ); vote.setTitle( title ); vote.append( "Yes", null ); vote.append( "No", null ); vote.setCommandListener( this ); Display.getDisplay(this).setCurrent(vote); } public void commandAction( Command c, Displayable d ) { if ( c == select && d == voteList ) { int index = voteList.getSelectedIndex(); print( ""+index ); showVote( index ); } } } Use BlueCove in this program. Server /* * To change this template, choose Tools | Templates * and open the template in the editor. */ package javaapplication4; import java.io.*; import java.util.concurrent.locks.Lock; import java.util.concurrent.locks.ReentrantLock; import javax.bluetooth.*; import javax.microedition.io.*; import javaapplication4.Connect; /** * * @author ????????????? */ public class SampleSPPServer { protected static int endToken = 255; private static Lock lock=new ReentrantLock(); private static StreamConnection conn = null; private static StreamConnectionNotifier streamConnNotifier = null; private void startServer() throws IOException{ //Create a UUID for SPP UUID uuid = new UUID("1101", true); //Create the service url String connectionString = "btspp://localhost:" + uuid +";name=Sample SPP Server"; //open server url StreamConnectionNotifier streamConnNotifier = (StreamConnectionNotifier)Connector.open( connectionString ); while ( true ) { Connect ct = new Connect( streamConnNotifier.acceptAndOpen() ); ct.getMessage(); } } /** * @param args the command line arguments */ public static void main(String[] args) { //display local device address and name try { LocalDevice localDevice = LocalDevice.getLocalDevice(); localDevice.setDiscoverable(DiscoveryAgent.GIAC); System.out.println("Name: "+localDevice.getFriendlyName()); } catch( Throwable e ) { e.printStackTrace(); } SampleSPPServer sampleSPPServer=new SampleSPPServer(); try { //start server sampleSPPServer.startServer(); } catch( IOException e ) { e.printStackTrace(); } } } Connect /* * To change this template, choose Tools | Templates * and open the template in the editor. */ package javaapplication4; import java.io.*; import java.util.concurrent.locks.Lock; import java.util.concurrent.locks.ReentrantLock; import javax.bluetooth.*; import javax.microedition.io.*; /** * * @author ????????????? */ public class Connect { private static DataInputStream in = null; private static DataOutputStream out = null; private static StreamConnection connection = null; private static Lock lock=new ReentrantLock(); public Connect( StreamConnection conn ) { connection = conn; } public synchronized void getMessage( ) { Thread t = new Thread() { public void run() { try { in = connection.openDataInputStream(); out = connection.openDataOutputStream(); String r = in.readUTF(); System.out.println("read:" + r); if ( r.equals( "c_getVotes" ) ) { out.writeUTF( "vote1" ); out.flush(); } if ( r.equals( "c_getVote_0" ) ) { out.writeUTF( "Vote1" ); out.flush(); } out.close(); in.close(); } catch (Throwable e) { } finally { if (in != null) { try { in.close(); } catch (IOException e) { } } try { connection.close(); } catch( IOException e ) { } } } }; t.start(); } }

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  • getaddrinfo appears to return different results between Windows and Ubuntu?

    - by MrDuk
    I have the following two sets of code: Windows #undef UNICODE #include <winsock2.h> #include <ws2tcpip.h> #include <stdio.h> // link with Ws2_32.lib #pragma comment (lib, "Ws2_32.lib") int __cdecl main(int argc, char **argv) { //----------------------------------------- // Declare and initialize variables WSADATA wsaData; int iResult; INT iRetval; DWORD dwRetval; argv[1] = "www.google.com"; argv[2] = "80"; int i = 1; struct addrinfo *result = NULL; struct addrinfo *ptr = NULL; struct addrinfo hints; struct sockaddr_in *sockaddr_ipv4; // struct sockaddr_in6 *sockaddr_ipv6; LPSOCKADDR sockaddr_ip; char ipstringbuffer[46]; DWORD ipbufferlength = 46; /* // Validate the parameters if (argc != 3) { printf("usage: %s <hostname> <servicename>\n", argv[0]); printf("getaddrinfo provides protocol-independent translation\n"); printf(" from an ANSI host name to an IP address\n"); printf("%s example usage\n", argv[0]); printf(" %s www.contoso.com 0\n", argv[0]); return 1; } */ // Initialize Winsock iResult = WSAStartup(MAKEWORD(2, 2), &wsaData); if (iResult != 0) { printf("WSAStartup failed: %d\n", iResult); return 1; } //-------------------------------- // Setup the hints address info structure // which is passed to the getaddrinfo() function ZeroMemory( &hints, sizeof(hints) ); hints.ai_family = AF_UNSPEC; hints.ai_socktype = SOCK_STREAM; // hints.ai_protocol = IPPROTO_TCP; printf("Calling getaddrinfo with following parameters:\n"); printf("\tnodename = %s\n", argv[1]); printf("\tservname (or port) = %s\n\n", argv[2]); //-------------------------------- // Call getaddrinfo(). If the call succeeds, // the result variable will hold a linked list // of addrinfo structures containing response // information dwRetval = getaddrinfo(argv[1], argv[2], &hints, &result); if ( dwRetval != 0 ) { printf("getaddrinfo failed with error: %d\n", dwRetval); WSACleanup(); return 1; } printf("getaddrinfo returned success\n"); // Retrieve each address and print out the hex bytes for(ptr=result; ptr != NULL ;ptr=ptr->ai_next) { printf("getaddrinfo response %d\n", i++); printf("\tFlags: 0x%x\n", ptr->ai_flags); printf("\tFamily: "); switch (ptr->ai_family) { case AF_UNSPEC: printf("Unspecified\n"); break; case AF_INET: printf("AF_INET (IPv4)\n"); sockaddr_ipv4 = (struct sockaddr_in *) ptr->ai_addr; printf("\tIPv4 address %s\n", inet_ntoa(sockaddr_ipv4->sin_addr) ); break; case AF_INET6: printf("AF_INET6 (IPv6)\n"); // the InetNtop function is available on Windows Vista and later // sockaddr_ipv6 = (struct sockaddr_in6 *) ptr->ai_addr; // printf("\tIPv6 address %s\n", // InetNtop(AF_INET6, &sockaddr_ipv6->sin6_addr, ipstringbuffer, 46) ); // We use WSAAddressToString since it is supported on Windows XP and later sockaddr_ip = (LPSOCKADDR) ptr->ai_addr; // The buffer length is changed by each call to WSAAddresstoString // So we need to set it for each iteration through the loop for safety ipbufferlength = 46; iRetval = WSAAddressToString(sockaddr_ip, (DWORD) ptr->ai_addrlen, NULL, ipstringbuffer, &ipbufferlength ); if (iRetval) printf("WSAAddressToString failed with %u\n", WSAGetLastError() ); else printf("\tIPv6 address %s\n", ipstringbuffer); break; case AF_NETBIOS: printf("AF_NETBIOS (NetBIOS)\n"); break; default: printf("Other %ld\n", ptr->ai_family); break; } printf("\tSocket type: "); switch (ptr->ai_socktype) { case 0: printf("Unspecified\n"); break; case SOCK_STREAM: printf("SOCK_STREAM (stream)\n"); break; case SOCK_DGRAM: printf("SOCK_DGRAM (datagram) \n"); break; case SOCK_RAW: printf("SOCK_RAW (raw) \n"); break; case SOCK_RDM: printf("SOCK_RDM (reliable message datagram)\n"); break; case SOCK_SEQPACKET: printf("SOCK_SEQPACKET (pseudo-stream packet)\n"); break; default: printf("Other %ld\n", ptr->ai_socktype); break; } printf("\tProtocol: "); switch (ptr->ai_protocol) { case 0: printf("Unspecified\n"); break; case IPPROTO_TCP: printf("IPPROTO_TCP (TCP)\n"); break; case IPPROTO_UDP: printf("IPPROTO_UDP (UDP) \n"); break; default: printf("Other %ld\n", ptr->ai_protocol); break; } printf("\tLength of this sockaddr: %d\n", ptr->ai_addrlen); printf("\tCanonical name: %s\n", ptr->ai_canonname); } freeaddrinfo(result); WSACleanup(); return 0; } Ubuntu /* ** listener.c -- a datagram sockets "server" demo */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/socket.h> #include <netinet/in.h> #include <arpa/inet.h> #include <netdb.h> #define MYPORT "4950" // the port users will be connecting to #define MAXBUFLEN 100 // get sockaddr, IPv4 or IPv6: void *get_in_addr(struct sockaddr *sa) { if (sa->sa_family == AF_INET) { return &(((struct sockaddr_in*)sa)->sin_addr); } return &(((struct sockaddr_in6*)sa)->sin6_addr); } int main(void) { int sockfd; struct addrinfo hints, *servinfo, *p; int rv; int numbytes; struct sockaddr_storage their_addr; char buf[MAXBUFLEN]; socklen_t addr_len; char s[INET6_ADDRSTRLEN]; memset(&hints, 0, sizeof hints); hints.ai_family = AF_UNSPEC; // set to AF_INET to force IPv4 hints.ai_socktype = SOCK_DGRAM; hints.ai_flags = AI_PASSIVE; // use my IP if ((rv = getaddrinfo(NULL, MYPORT, &hints, &servinfo)) != 0) { fprintf(stderr, "getaddrinfo: %s\n", gai_strerror(rv)); return 1; } // loop through all the results and bind to the first we can for(p = servinfo; p != NULL; p = p->ai_next) { if ((sockfd = socket(p->ai_family, p->ai_socktype, p->ai_protocol)) == -1) { perror("listener: socket"); continue; } if (bind(sockfd, p->ai_addr, p->ai_addrlen) == -1) { close(sockfd); perror("listener: bind"); continue; } break; } if (p == NULL) { fprintf(stderr, "listener: failed to bind socket\n"); return 2; } freeaddrinfo(servinfo); printf("listener: waiting to recvfrom...\n"); addr_len = sizeof their_addr; if ((numbytes = recvfrom(sockfd, buf, MAXBUFLEN-1 , 0, (struct sockaddr *)&their_addr, &addr_len)) == -1) { perror("recvfrom"); exit(1); } printf("listener: got packet from %s\n", inet_ntop(their_addr.ss_family, get_in_addr((struct sockaddr *)&their_addr), s, sizeof s)); printf("listener: packet is %d bytes long\n", numbytes); buf[numbytes] = '\0'; printf("listener: packet contains \"%s\"\n", buf); close(sockfd); return 0; } When I attempt www.google.com, I don't get the ipv6 socket returned on Windows - why is this? Outputs: (ubuntu) caleb@ub1:~/Documents/dev/cs438/mp0/MP0$ ./a.out www.google.com IP addresses for www.google.com: IPv4: 74.125.228.115 IPv4: 74.125.228.116 IPv4: 74.125.228.112 IPv4: 74.125.228.113 IPv4: 74.125.228.114 IPv6: 2607:f8b0:4004:803::1010 Outputs: (win) Calling getaddrinfo with following parameters: nodename = www.google.com servname (or port) = 80 getaddrinfo returned success getaddrinfo response 1 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.114 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 2 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.115 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 3 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.116 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 4 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.112 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 5 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.113 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null)

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  • AVCHD MTS h264 1080p file with choppy playback in Linux

    - by marc
    When I'm trying play video files from my camera: Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) Input #0, mpegts, from '00027.MTS': Duration: 00:00:38.88, start: 2.884289, bitrate: 16945 kb/s Program 1 Stream #0.0[0x1011]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s … on my Linux computer (Ubuntu 12.04), I get choppy playback. It's completly unusable... I tried: Totem VLC mplayer The result is always same issue. I sent the same video file to a friend who has ubuntu 10.04 to test, and he also has the same issue. He has Windows 7, and confirms that on Windows, the video work well. I have an Intel® Core™2 CPU 6300 @ 1.86GHz × 2 with GF 9600 GT, with closed NVIDIA drivers. This is not any kind of issue with big files playing slow from an HDD issue. I have an SSD drive! I spent the last days and nights, trying hundreds of commands for ffmpeg, handbrake, mencoder... Any of them won't let me create a file with enough quality. I downloaded few movies from YouTube in 1080p, and playback worked well without any big pixels and choppiness. I would like have highest possible quality, I will put following files onto a Blu-ray disk so I don't need to compress them to get a smaller size. I just want smoth playback on my Linux box. On Windows, the same file is working well.

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  • Looking for an application to record audio and video on a linux "embedded" device

    - by Luke404
    I am working with a linux x86 device with limited CPU resources (as a prototype we just use a pentium-m netbook). We'd like to record video from one V4L2 device (we'll probably end up using just USB Video Class devices like all modern webcams) and one audio stream from an ALSA source. The thing will not have screen and keyboard, and obviously no X11 environment. Goals are: do as little work as possible to cope with little cpu resources - for example I'd like to record video in the native MJPEG I get out of the UVC devices encoding audio to MPEG3 Layer-2 (aka mp2) is ok since it let us save a lot of space (compared to raw pcm samples) and does use little cpu power I don't mind loosing some video frames here and there (UVC devices do that) as long as I can get audio and video streams syncronized not require user input to start the thing (a python script takes care of initialization, startup, shutdown, etc...) be able to open the resulting files for postprocessing without too much effort (ie, if mplayer or vlc can play it, it's fine) So far the only app I found that could be started from command line and record V4L2 video + ALSA audio is mencoder but I'm having some difficulties with it. It should be able to do that but I cannot record audio and video together - just one of the two. And if I use two different processes to record to two different files I have no means to get them in sync (audio is more or less always correct, but video framerate will vary over time and it seems to lack timestamps to correctly play it back to the correct time). Long story short, how do you record an unconverted MJPEG stream (from an UVC device) and an audio stream (from an ALSA device, possibly encoding to any standard format) using a command line tool, to a single file (MPEG or any other container), keeping audio and video in sync?

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  • Cannot establish XMPP server-to-server connection to gmail

    - by v_2e
    My jabber-server fails to connect to gmail.com giving the error: outgoing s2s stream myserver.com.ua-bot.talk.google.com closed: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) I am using the Prosody XMPP server. It works just fine with other jabber-servers I tested so far (e.g. jabber.ru). However, when some of my clients tries to add a gmail contact to his contact-list, the subscription request lasts forever, and the Prosody gives the following sequence of messages in its log: Oct 21 22:57:16 s2sout95897f8 info Beginning new connection attempt to gmail.com ([173.194.70.125]:5269) Oct 21 22:57:16 s2sout95897f8 info sent dialback key on outgoing s2s stream Oct 21 22:57:16 s2sout95897f8 info Session closed by remote with error: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) Oct 21 22:57:16 s2sout95897f8 info outgoing s2s stream myserver.com.ua->gmail.com closed: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) Oct 21 22:57:16 s2sout95897f8 info sending error replies for 2 queued stanzas because of failed outgoing connection to gmail.com Here for the domain name of my server I use myserver.com.ua I found a similar problem described in this thread, but there is no detailed description of the solution there. As for the Google services, I did have a google account where I added the domain name under question to the Webmasters tools page. However, I deleted my account long ago, so now it is unclear, how any of the Google services can relate to my domain name. So my question is: What is the real cause of this problem (my jabber-server configuration or imaginary Google account or something else) and how can I make my Prosody server connect to gmail.com jabber service?

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  • Why is my mono/XSP site loading slow?

    - by acidzombie24
    I have two sites on the same server. One is loading perfectly and incredibly fast. The other is an equally complex site except a bit less javascript and 0 images. Its taking several seconds to load and there is a 1 in 3 chance that i get a Http500 error. WTF I grabbed the lastest 2.6.? version of mono, mod_mono and xsp (libgdiplus-2.6.7, xsp-2.6.5, mod_mono-2.6.3 and mono 2.6.7) This is whats in apache error.log [Mon Jan 03 19:33:40 2011] [error] (70014)End of file found: read_data failed [Mon Jan 03 19:33:40 2011] [error] Command stream corrupted, last command was 1 [Mon Jan 03 19:34:52 2011] [error] (70014)End of file found: read_data failed [Mon Jan 03 19:34:52 2011] [error] (70014)End of file found: read_data failed [Mon Jan 03 19:34:52 2011] [error] Command stream corrupted, last command was 1 [Mon Jan 03 19:34:52 2011] [error] Command stream corrupted, last command was 1 this is the page error Internal Server Error The server encountered an internal error or misconfiguration and was unable to complete your request. Please contact the server administrator, webmaster@localhost and inform them of the time the error occurred, and anything you might have done that may have caused the error. More information about this error may be available in the server error log. Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny9 with Suhosin-Patch mod_mono/2.6.3 Server

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  • ffmpeg conversion problem

    - by user33126
    installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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  • ffmpeg conversion problem

    - by Elamurugan
    installed ffmpeg and it shows version and all correctly. but even info ffmpeg command itself shows ffmpeg -i Alice_In_Wonderland.mp4 gives messgae like FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --extra-cflags=-fPIC --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:11:04, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 1 codec frame rate differs from container frame rate: 49.93 (9986/200) - 49.92 (599/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Alice_In_Wonderland.mp4': Duration: 00:01:39.65, start: 0.000000, bitrate: 542 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Stream #0.1(und): Video: h264, yuv420p, 480x270, 49.92 tbr, 24.96 tbn, 49.93 tbc At least one output file must be specified Please tell me whats the problem

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  • Cannot establish XMPP server-to-server connection with gmail

    - by v_2e
    My jabber-server fails to connect to gmail.com giving the error: outgoing s2s stream myserver.com.ua-bot.talk.google.com closed: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) I am using the Prosody XMPP server. It works just fine with other jabber-servers I tested so far (e.g. jabber.ru). However, when some of my clients tries to add a gmail contact to his contact-list, the subscription request lasts forever, and the Prosody gives the following sequence of messages in its log: Oct 21 22:57:16 s2sout95897f8 info Beginning new connection attempt to gmail.com ([173.194.70.125]:5269) Oct 21 22:57:16 s2sout95897f8 info sent dialback key on outgoing s2s stream Oct 21 22:57:16 s2sout95897f8 info Session closed by remote with error: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) Oct 21 22:57:16 s2sout95897f8 info outgoing s2s stream myserver.com.ua->gmail.com closed: undefined-condition (myserver.com.ua is a Google Apps Domain with Talk service enabled.) Oct 21 22:57:16 s2sout95897f8 info sending error replies for 2 queued stanzas because of failed outgoing connection to gmail.com Here for the domain name of my server I use myserver.com.ua I found a similar problem described in this thread, but there is no detailed description of the solution there. As for the Google services, I did have a google account where I added the domain name under question to the Webmasters tools page. However, I deleted my account long ago, so now it is unclear, how any of the Google services can relate to my domain name. So my question is: What is the real cause of this problem (my jabber-server configuration or imaginary Google account or something else) and how can I make my Prosody server connect to gmail.com jabber service?

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  • ffmpeg: video file played OK on Ubuntu, but no sound on XP

    - by Andy Le
    I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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