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  • ALSA Mutes Incorrect Audio Device on Headphone Event

    - by Gardiner
    I know what the problem is, I just don't know how to prevent it. When I plug in my headphones, ALSA automatically mutes (and sets the volume to 0) the 'Speaker' entry in alsamixer. If I go in and manually unmute (and turn the volume up) I have sound from my headphones. Question is, how do I change this behavior? I want it to adjust the volume to half, but not mute or set the volume to 0. I'm using Ubuntu Gnome 13.04 Output of amixer scontents Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] This answer solved my issue: How can I get sound on headphones without switching back to 'Speakers' manually?

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  • USB audio device not detected [solved]

    - by user199052
    I've been having problems with USB sound for some time (it used to work about 6 months ago). I find that any USB sound device that I connect is not recognised, and is not listed by aplay -l I've tried disabling pulseaudio. I've tried purging and re-installing alsa-base, pulseaudio, and pavucontrol. To no avail. I'm using 12.04 LTS. I found the solution: 12.04 does not have the needed kernel module snd-usb-hiface this is introduced in the 3.11.1 kernel and installing that on 12.04 gives USB sound for my equipment. How to install the 3.11.1 kernel is given here:- installing 3.11.1 kernel

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  • Comet VS Ajax polling

    - by xRobot
    I need to create a chat like facebook chat. With Comet I need more memory to keep the connection. With Ajax polling there is a latency problem if I send request every 3-4 seconds. So... If the latency ( 3-4 seconds ) doesn't matter, Is Ajax Polling better for my case ?

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  • How to send audio stream via UDP in java?

    - by Nob Venoda
    Hi to all :) I have a problem, i have set MediaLocator to microphone input, and then created Player. I need to grab that sound from the microphone, encode it to some lower quality stream, and send it as a datagram packet via UDP. Here's the code, i found most of it online and adapted it to my app: public class AudioSender extends Thread { private MediaLocator ml = new MediaLocator("javasound://44100"); private DatagramSocket socket; private boolean transmitting; private Player player; TargetDataLine mic; byte[] buffer; private AudioFormat format; private DatagramSocket datagramSocket(){ try { return new DatagramSocket(); } catch (SocketException ex) { return null; } } private void startMic() { try { format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 8000.0F, 16, 2, 4, 8000.0F, true); DataLine.Info info = new DataLine.Info(TargetDataLine.class, format); mic = (TargetDataLine) AudioSystem.getLine(info); mic.open(format); mic.start(); buffer = new byte[1024]; } catch (LineUnavailableException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } private Player createPlayer() { try { return Manager.createRealizedPlayer(ml); } catch (IOException ex) { return null; } catch (NoPlayerException ex) { return null; } catch (CannotRealizeException ex) { return null; } } private void send() { try { mic.read(buffer, 0, 1024); DatagramPacket packet = new DatagramPacket( buffer, buffer.length, InetAddress.getByName(Util.getRemoteIP()), 91); socket.send(packet); } catch (IOException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void run() { player = createPlayer(); player.start(); socket = datagramSocket(); transmitting = true; startMic(); while (transmitting) { send(); } } public static void main(String[] args) { AudioSender as = new AudioSender(); as.start(); } } And only thing that happens when I run the receiver class, is me hearing this Player from the sender class. And I cant seem to see the connection between TargetDataLine and Player. Basically, I need to get the sound form player, and somehow convert it to bytes[], therefore I can sent it as datagram. Any ideas? Everything is acceptable, as long as it works :)

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  • Streaming Audio from A URL in Android using MediaPlayer?

    - by Sena Gbeckor-Kove
    Hi, I've been trying to stream mp3's over http using Android's built in MediaPlayer class. The documentation would suggest to me that this should be as easy as : MediaPlayer mp = new MediaPlayer(); mp.setDataSource(URL_OF_FILE); mp.prepare(); mp.start(); However I am getting the following repeatedly. I have tried different URLs as well. Please don't tell me that streaming doesn't work on mp3's. E/PlayerDriver( 31): Command PLAYER_SET_DATA_SOURCE completed with an error or info PVMFErrNotSupported W/PlayerDriver( 31): PVMFInfoErrorHandlingComplete E/MediaPlayer( 198): error (1, -4) E/MediaPlayer( 198): start called in state 0 E/MediaPlayer( 198): error (-38, 0) E/MediaPlayer( 198): Error (1,-4) E/MediaPlayer( 198): Error (-38,0) Any help much appreciated, thanks S

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  • Is the Finch audio library for iPhone capable of doing this?

    - by mystify
    I need to: - start / stop sounds with lengths between 0.1 and 10 seconds - change the playback volume I want to / would like to / would be nice to have to: - change the playback speed - change the playback pitch / frequency - pause an sound and resume playing it later - play a sound backwards Is Finch my best friend here?

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  • jsp chatting with ajax

    - by paramesh
    Hi sirs, I am writing one web chat program using AJAX (a little bit). It is working when both users open a chat page, but I want to open a window when one user send data to others.

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  • how to get audio frequency data from a wave file?

    - by potlee
    I want to build a speech recognition engine in ruby. I know i'll never get there, doing it just for fun. I need to get data for the frequencies of the sound stored in a wav file to compare with data i already have of different sounds that i want to recognize. I will write the code in ruby but i dont think there are any libraries for this written in ruby, they would be too slow if there were any anyway. The good thing about ruby is I'll be able to use libraries for .net via IronRuby or Java via Jruby. How can i get the frequency data?

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  • Silverlight Chat WrapPanel Crash / Bug

    - by Matt
    I've been given the task to create a simple Silverlight chat box for two people. My control must adhere to the following requirements Scrollable Text must wrap if it's too long When a new item / message is added it must scroll that item into view Now I've successfully made a usercontrol to meet these requirements, but I've run into a possible bug / crash that I can't for the life of me fix. I'm looking for either a fix to the bug, or a different approach to creating a scrollable chat control. Here's the code I've been using. We'll start with my XAML for the chat window <ListBox x:Name="lbChatHistory" Grid.Row="0" Grid.Column="0" Grid.ColumnSpan="2" ScrollViewer.VerticalScrollBarVisibility="Visible" ScrollViewer.HorizontalScrollBarVisibility="Disabled" > <ListBox.ItemTemplate> <DataTemplate> <Grid Background="Beige"> <Grid.ColumnDefinitions> <ColumnDefinition Width="70"></ColumnDefinition> <ColumnDefinition Width="Auto"></ColumnDefinition> </Grid.ColumnDefinitions> <TextBlock x:Name="lblPlayer" Foreground="{Binding ForeColor}" Text="{Binding Player}" Grid.Column="0"></TextBlock> <ContentPresenter Grid.Column="1" Width="200" Content="{Binding Message}" /> </Grid> </DataTemplate> </ListBox.ItemTemplate> </ListBox> The idea is to add a new Item to the listbox. The Item (as layed out in the XAML) is a simple 2 column grid. One column for the username, and one column for the message. Now the "items" that I add to the ListBox is a custom class. It has three properties (Player, ForeColor, and Message) that I using binding on within my XAML Player is a string of the current user to display. ForeColor is just a foreground color preference. It helps distinguish the difference between messages. Message is a WrapPanel. I programmatically break the supplied string on the white space for each word. Then for each word, I add a new TextBlock element to the WrapPanel Here is the custom class. public class ChatMessage :DependencyObject, INotifyPropertyChanged { public event PropertyChangedEventHandler PropertyChanged; public static DependencyProperty PlayerProperty = DependencyProperty.Register( "Player", typeof( string ), typeof( ChatMessage ), new PropertyMetadata( new PropertyChangedCallback( OnPlayerPropertyChanged ) ) ); public static DependencyProperty MessageProperty = DependencyProperty.Register( "Message", typeof( WrapPanel ), typeof( ChatMessage ), new PropertyMetadata( new PropertyChangedCallback( OnMessagePropertyChanged ) ) ); public static DependencyProperty ForeColorProperty = DependencyProperty.Register( "ForeColor", typeof( SolidColorBrush ), typeof( ChatMessage ), new PropertyMetadata( new PropertyChangedCallback( OnForeColorPropertyChanged ) ) ); private static void OnForeColorPropertyChanged( DependencyObject d, DependencyPropertyChangedEventArgs e ) { ChatMessage c = d as ChatMessage; c.ForeColor = ( SolidColorBrush ) e.NewValue; } public ChatMessage() { Message = new WrapPanel(); ForeColor = new SolidColorBrush( Colors.White ); } private static void OnMessagePropertyChanged( DependencyObject d, DependencyPropertyChangedEventArgs e ) { ChatMessage c = d as ChatMessage; c.Message = ( WrapPanel ) e.NewValue; } private static void OnPlayerPropertyChanged( DependencyObject d, DependencyPropertyChangedEventArgs e ) { ChatMessage c = d as ChatMessage; c.Player = e.NewValue.ToString(); } public SolidColorBrush ForeColor { get { return ( SolidColorBrush ) GetValue( ForeColorProperty ); } set { SetValue( ForeColorProperty, value ); if(PropertyChanged != null) PropertyChanged(this, new PropertyChangedEventArgs( "ForeColor" )); } } public string Player { get { return ( string ) GetValue( PlayerProperty ); } set { SetValue( PlayerProperty, value ); if ( PropertyChanged != null ) PropertyChanged( this, new PropertyChangedEventArgs( "Player" ) ); } } public WrapPanel Message { get { return ( WrapPanel ) GetValue( MessageProperty ); } set { SetValue( MessageProperty, value ); if ( PropertyChanged != null ) PropertyChanged( this, new PropertyChangedEventArgs( "Message" ) ); } } } Lastly I add my items to the ListBox. Here's the simple method. It takes the above ChatMessage class as a parameter public void AddChatItem( ChatMessage msg ) { lbChatHistory.Items.Add( msg ); lbChatHistory.ScrollIntoView( msg ); } Now I've tested this and it all works. The problem I'm getting is when I use the scroll bar. You can scroll down using the side scroll bar or arrow keys, but when you scroll up Silverlight crashes. FireBug returns a ManagedRuntimeError #4004 with a XamlParseException. I'm soo close to having this control work, I can taste it! Any thoughts on what I should do or change? Is there a better approach than the one I've taken? Thanks in advance. UPDATE I've found an alternative solution using a ScrollViewer and an ItemsControl instead of a ListBox control. For the most part it's stable.

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  • Organisation d'une discussion "live" autour de l'emploi sur le chat de Developpez.com, le mardi 8 février

    Chers membres du forum, La période des stages approche,et certains d'entre vous ont sûrement des questions sur leur CV, leur lettre de motivation ou tout simplement le monde du travail. Le chat de Developpez.com vous propose donc le mardi 8 février de faire une session Emploi. L'objectif principal est de se réunir de façon informelle autour des questions de l'emploi, de partager ses expériences. Pour le fonctionnement, rien de plus simple :Si vous ne connaissez pas le chat, merci tout d'abord de lire l'aide ; Connectez-vous au chat avec...

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  • How to route KVM virtual machine audio to Ubuntu 11.10 host using virt-manager?

    - by iGadget
    I've been using KVM in combination with Virt-Manager and Remmina at a fair success up until now. The issue I need to solve now is to get audio from a virtualized Windows XP and make it audible on the Ubuntu 11.10 host. Remmina / RDP works for 'simple' audio (system sounds and such), but when the source gets trickier (e.g. Flash audio), Remmina / RDP messes up. So I figured I'd just connect to the machine directly using Virt-Manager. Unfortunately, it seems that even though I have successfully configured the AC97 audio device on WinXP, it's unable to get it's output to the Ubuntu host. This is probably because Virt-Manager uses VNC (and AFAIK, VNC doesn't transport audio). Does anyone know if there is a solution to fix this? I've heard of Spice, but the installation required so much voodoo last time I checked, I figured I'd let that solution boil to maturity a little longer ;) But perhaps there are other options I haven't thought of yet (which don't require switching to VirtualBox / VMware)...

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  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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  • Why is a FLAC encoded from a decoded MP3 bigger than the MP3?

    - by Ryan Thompson
    To be more precise than in the title, suppose I have a MP3 file that is 320 kbps. If I decompress it, then logically, all the data except for roughly 320 kilobits out of each second of audio should be redundant data, able to be compressed away. So, when I encode the decompressed file to FLAC, or any other lossless codec, why is it so much larger? On a related note, is it theoretically possible to losslessly recover the source mp3 audio from a decompressed wav? (I know the mp3 itself is lossy. I'm asking if it's possible to re-encode without any further loss.) EDIT: Let me clarify the related question, and the rationale behind it. Suppose I have a wav that was decompressed from an MP3 file (and assume I don't have the mp3 itself for some reason). If I don't want to lose any more quality, I can re-encode it with FLAC or any other lossless encoder and get a larger file just to maintain the same quality. Or, I can re-encode it to mp3 again and get the same size as the original but lose more data. Obviously, neither of these cases is ideal. I can either have the original size or the original quality, but not both (I mean the quality of the original mp3, not the original lossless source). My question is: Can we get both? Is it theoretically possible to recover the lossy compressed data from the lossy decompressed data, without losing even more? If it is possible, I could imagine a lossless compression algorithm that compresses the audio with FLAC. Then it also scans the audio for any signs of previous lossy compression, and if detected, recompresses it losslessly to the original lossy file. Then it keeps whichever file is smaller.

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