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  • How to pipe internet radio into a tuner?

    - by JW
    UPDATE: Thanks everyone for the ideas! This was an area I knew very little about but now I can talk with a little more expertise about it. Much appreciated! Visited my dad this weekend and he wants to pipe some internet radio he's found down to a tuner on quite a distance away in the house. He uses computers for only very basic things: e-mail, getting the Post crossword, checking Yahoo!, checking recipes, etc. There's currently one computer in the house (no router). My initial suggestion (without any research whatsoever) was to get a wireless router and a netbook for downstairs near the tuner, but he initially wasn't too keen about having another computer down there. Anyway, is there any computer hardware that could magically pipe the audio output from the computer down to one set of (RCA) audio inputs on the tuner? Wireless isn't necessary but it probably would be easier. Anyway, thanks for your suggestions! UPDATE Thanks everyone! Voted up all of your suggestions now that I have 15 rep. Much appreciated.

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  • Image compression and thumbnail creation for PHP

    - by Saif Bechan
    I need some PHP classes that deal with image processing in a good manner. I have made a thumbnail creator myself but the end result quality is just horrible. Is it also possible to let PHP convert and save all images to one type. For example take an image(jpg,png,gif), compress it, resize it, and save as png. Can anyone recommend some good classes for this.

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  • How can I control which sound card Ubuntu uses for playback?

    - by GorillaSandwich
    I am dual-booting Ubuntu 9.04 and Windows XP but am new to Ubuntu. In Windows, I use an M-Audio Audiophile 2496 sound card for recording (because it has RCA input jacks for my mixer), but I don't use it for playback (because my speakers use a 1/8 inch jack); instead, I use the motherboard's built-in sound card. I tried to recreate this arrangement in Ubuntu, but despite selecting the built-in card for all playback under System > Preferences > Sound, I still have inconsistent results. Rhythmbox plays back through the integrated card, but Flash content in the browser and games in the OS send their audio to the Audiophile card. I have seen recommendations to use a program called "Jack" to control this, but I installed it and found it baffling. How can I control which card is used for playback, other than disabling one card (as I discovered how to do and explain below)? Also, is there a GUI for disabling hardware, or is it necessary to edit a configuration file?

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  • LZMA compression settings details

    - by Shadi
    Hi everyone, I really need to know what each lzma parameter (mf, fb, lp, ...) means. I could not find any good documentation in the internet. I need details of this algorithm. the most detailed one is: http://www.bugaco.com/7zip/MANUAL/switches/method.htm I would appreciate any help. Best wishes, Shadi.

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  • Uploadify and Image Compression

    - by Ilya Biryukov
    Hi, I am using Uploadify on one of my client's web sites to allow them to upload a large amount of pictures at once to their photo gallery. I am seeing issues lately. They seem to upload large photographs (3 MB and above). I am wondering, is it possible to compress (reduce their size) on the client side, instead of doing it on the server (just like facebook does it). I know I could easily do it on the server, but I am working on another project right now, where I am expecting a large flow of photo uploads. It would require significant amount of CPU time to process them all. So I thought, I'd ask about the client side processing. Thanks.

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  • Great guide for JavaScript GZIP compression in IIS?

    - by Django Reinhardt
    Hi there, we're looking to compress our gargantuan JavaScript files with GZip to speed up the page loads of our site. I know this can be done through IIS, but I can't seem to find a simple step-by-step guide on how to implement it. If someone could point me towards such a guide, I'd really appreciate it. I've never done this before, so it would need to be quite basic. We're running IIS7.5 on Windows Server 2008 R2. Your time is much appreciated.

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  • bitmap compression

    - by seddik
    hi programmers i'm trying to send a bitmap screenshot over network , so i need to compress it before sending it, so if anyone have a librery or methode just tell me , thanx a lot

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  • WebClient and Gzip compression is faster?

    - by Yozer
    I writting an application which is using WebClient class. Adding something like that: ExC.Headers.Add("Accept-Encoding: gzip, deflate"); where ExC is: class ExWebClient1 : WebClient { protected override WebRequest GetWebRequest(Uri address) { HttpWebRequest request = (HttpWebRequest)base.GetWebRequest(address); request.AutomaticDecompression = DecompressionMethods.GZip | DecompressionMethods.Deflate; return request; } } It will be a diffrence in speed when i will be using encoded response?

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  • Database Compression in Python

    - by user551832
    I have hourly logs like user1:joined user2:log out user1:added pic user1:added comment user3:joined I want to compress all the flat files down to one file. There are around 30 million users in the logs and I just want the latest user log for all the logs. My end result is I want to have a log look like user1:added comment user2:log out user3:joined Now my first attempt on a small scale was to just do a dict like log['user1'] = "added comment" Will doing a dict of 30 million key/val pairs have a giant memory footprint.. Or should I use something like sqllite to store them.. then just put the contents of the sqllite table back into a file?

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  • Distributing cpu-bound compression jobs to multiple computers?

    - by barnaby
    The other day I needed to archive a lot of data on our network and I was frustrated I had no immediate way to harness the power of multiple machines to speed-up the process. I understand that creating a distributed job management system is a leap from a command-line archiving tool. I'm now wondering what the simplest solution to this type of distributed performance scenario could be. Would a custom tool always be a requirement or are there ways to use standard utilities and somehow distribute their load transparently at a higher level? Thanks for any suggestions.

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  • Compression Program in C

    - by Delandilon
    I want to compress a series of characters. For example if i type Input : FFFFFBBBBBBBCCBBBAABBGGGGGSSS (27 x 8 bits = 216 bits) Output: F5B7C2B3A2B2G5S3 (14 x 8 bits = 112bits) So far this is what i have, i can count the number of Characters in the Array. But the most important task is to count them in the same sequence. I can't seem to figure that out :( Ive stared doing C just a few weeks back, i have knowledge on Array, pointers, ASCII value but in any case can't seem to count these characters in a sequence. Ive try a bit of everything. This approach is no good but it the closest i came to it. #include <stdio.h> #include <conio.h> int main() { int charcnt=0,dotcnt=0,commacnt=0,blankcnt=0,i, countA, countB; char str[125]; printf("*****String Manipulations*****\n\n"); printf("Enter a string\n\n"); scanf("%[^'\n']s",str); printf("\n\nEntered String is \" %s \" \n",str); for(i=0;str[i]!='\0';i++) { // COUNTING EXCEPTION CHARS if(str[i]==' ') blankcnt++; if(str[i]=='.') dotcnt++; if(str[i]==',') commacnt++; if (str[i]=='A' || str[i]=='a') countA++; if (str[i]=='B' || str[i]=='b') countA++; } //PRINT RESULT OF COUNT charcnt=i; printf("\n\nTotal Characters : %d",charcnt); printf("\nTotal Blanks : %d",blankcnt); printf("\nTotal Full stops : %d",dotcnt); printf("\nTotal Commas : %d\n\n",commacnt); printf("A%d\n", countA); }

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  • How to send audio data from Java Applet to Rails controller

    - by cooldude
    Hi, I have to send the audio data in byte array obtain by recording from java applet at the client side to rails server at the controller in order to save. So, what encoding parameters at the applet side be used and in what form the audio data be converted like String or byte array so that rails correctly recieve data and then I can save that data at the rails in the file. As currently the audio file made by rails controller is not playing. It is the following ERROR : LAVF_header: av_open_input_stream() failed while playing with the mplayer. Here is the Java Code: package networksocket; import java.util.logging.Level; import java.util.logging.Logger; import javax.swing.JApplet; import java.net.*; import java.io.*; import java.awt.event.*; import java.awt.*; import java.sql.*; import javax.swing.*; import javax.swing.border.*; import java.awt.*; import java.util.Properties; import javax.swing.plaf.basic.BasicSplitPaneUI.BasicHorizontalLayoutManager; import sun.awt.HorizBagLayout; import sun.awt.VerticalBagLayout; import sun.misc.BASE64Encoder; /** * * @author mukand */ public class Urlconnection extends JApplet implements ActionListener { /** * Initialization method that will be called after the applet is loaded * into the browser. */ public BufferedInputStream in; public BufferedOutputStream out; public String line; public FileOutputStream file; public int bytesread; public int toread=1024; byte b[]= new byte[toread]; public String f="FINISH"; public String match; public File fileopen; public JTextArea jTextArea; public Button refreshButton; public HttpURLConnection urlConn; public URL url; OutputStreamWriter wr; BufferedReader rd; @Override public void init() { // TODO start asynchronous download of heavy resources //textField= new TextField("START"); //getContentPane().add(textField); JPanel p = new JPanel(); jTextArea= new JTextArea(1500,1500); p.setLayout(new GridLayout(1,1, 1,1)); p.add(new JLabel("Server Details")); p.add(jTextArea); Container content = getContentPane(); content.setLayout(new GridBagLayout()); // Used to center the panel content.add(p); jTextArea.setLineWrap(true); refreshButton = new java.awt.Button("Refresh"); refreshButton.reshape(287,49,71,23); refreshButton.setFont(new Font("Dialog", Font.PLAIN, 12)); refreshButton.addActionListener(this); add(refreshButton); Properties properties = System.getProperties(); properties.put("http.proxyHost", "netmon.iitb.ac.in"); properties.put("http.proxyPort", "80"); } @Override public void actionPerformed(ActionEvent e) { try { url = new URL("http://localhost:3000/audio/audiorecieve"); urlConn = (HttpURLConnection)url.openConnection(); //String login = "mukandagarwal:rammstein$"; //String encodedLogin = new BASE64Encoder().encodeBuffer(login.getBytes()); //urlConn.setRequestProperty("Proxy-Authorization",login); urlConn.setRequestMethod("POST"); // urlConn.setRequestProperty("Content-Type", //"application/octet-stream"); //urlConn.setRequestProperty("Content-Type","audio/mpeg");//"application/x-www- form-urlencoded"); //urlConn.setRequestProperty("Content-Type","application/x-www- form-urlencoded"); //urlConn.setRequestProperty("Content-Length", "" + // Integer.toString(urlParameters.getBytes().length)); urlConn.setRequestProperty("Content-Language", "UTF-8"); urlConn.setDoOutput(true); urlConn.setDoInput(true); byte bread[]=new byte[2048]; int iread; char c; String data=URLEncoder.encode("key1", "UTF-8")+ "="; //String data="key1="; FileInputStream fileread= new FileInputStream("//home//mukand//Hellion.ogg");//Dogs.mp3");//Desktop//mausam1.mp3"); while((iread=fileread.read(bread))!=-1) { //data+=(new String()); /*for(int i=0;i<iread;i++) { //c=(char)bread[i]; System.out.println(bread[i]); }*/ data+= URLEncoder.encode(new String(bread,iread), "UTF-8");//new String(new String(bread));// // data+=new String(bread,iread); } //urlConn.setRequestProperty("Content-Length",Integer.toString(data.getBytes().length)); System.out.println(data); //data+=URLEncoder.encode("mukand", "UTF-8"); //data += "&" + URLEncoder.encode("key2", "UTF-8") + "=" + URLEncoder.encode("value2", "UTF-8"); //data="key1="; wr = new OutputStreamWriter(urlConn.getOutputStream());//urlConn.getOutputStream(); //if((iread=fileread.read(bread))!=-1) // wr.write(bread,0,iread); wr.write(data); wr.flush(); fileread.close(); jTextArea.append("Send"); // Get the response rd = new BufferedReader(new InputStreamReader(urlConn.getInputStream())); while ((line = rd.readLine()) != null) { jTextArea.append(line); } wr.close(); rd.close(); //jTextArea.append("click"); } catch (MalformedURLException ex) { Logger.getLogger(Urlconnection.class.getName()).log(Level.SEVERE, null, ex); } catch (IOException ex) { Logger.getLogger(Urlconnection.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void start() { } @Override public void stop() { } @Override public void destroy() { } // TODO overwrite start(), stop() and destroy() methods } Here is the Rails controller function for recieving: def audiorecieve puts "///////////////////////////////////////******RECIEVED*******////" puts params[:key1]#+" "+params[:key2] data=params[:key1] #request.env('RAW_POST_DATA') file=File.new("audiodata.ogg", 'w') file.write(data) file.flush file.close puts "////**************DONE***********//////////////////////" end Please reply quickly

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  • audio onprogress in chrome not working

    - by user351709
    Hi I am having a problem getting onprogress event for the audio tag working on chrome. it seems to work on fire fox. http://www.scottandrew.com/pub/html5audioplayer/ works on chrome but there is no progress bar update. When I copy the code and change the src to a .wav file and run it on fire fox it works perfectly. <style type="text/css"> #content { clear:both; width:60%; } .player_control { float:left; margin-right:5px; height: 20px; } #player { height:22px; } #duration { width:400px; height:15px; border: 2px solid #50b; } #duration_background { width:400px; height:15px; background-color:#ddd; } #duration_bar { width:0px; height:13px; background-color:#bbd; } #loader { width:0px; height:2px; } .style1 { height: 35px; } </style> <script type="text/javascript"> var audio_duration; var audio_player; function pageLoaded() { audio_player = $("#aplayer").get(0); //get the duration audio_duration = audio_player.duration; $('#totalTime').text(formatTimeSeconds(audio_player.duration)); //set the volume } function update(){ //get the duration of the player dur = audio_player.duration; time = audio_player.currentTime; fraction = time/dur; percent = (fraction*100); wrapper = document.getElementById("duration_background"); new_width = wrapper.offsetWidth*fraction; document.getElementById("duration_bar").style.width = new_width + "px"; $('#currentTime').text(formatTimeSeconds(audio_player.currentTime)); $('#totalTime').text(formatTimeSeconds(audio_player.duration)); } function formatTimeSeconds(time) { var minutes = Math.floor(time / 60); var seconds = "0" + (Math.floor(time) - (minutes * 60)).toString(); if (isNaN(minutes) || isNaN(seconds)) { return "0:00"; } var Strseconds = seconds.substr(seconds.length - 2); return minutes + ":" + Strseconds; } function playClicked(element){ //get the state of the player if(audio_player.paused) { audio_player.play(); newdisplay = "||"; }else{ audio_player.pause(); newdisplay = ">"; } $('#totalTime').text(formatTimeSeconds(audio_player.duration)); element.value = newdisplay; } function trackEnded(){ //reset the playControl to 'play' document.getElementById("playControl").value=">"; } function durationClicked(event){ //get the position of the event clientX = event.clientX; left = event.currentTarget.offsetLeft; clickoffset = clientX - left; percent = clickoffset/event.currentTarget.offsetWidth; duration_seek = percent*audio_duration; document.getElementById("aplayer").currentTime=duration_seek; } function Progress(evt){ $('#progress').val(Math.round(evt.loaded / evt.total * 100)); var width = $('#duration_background').css('width') $('#loader').css('width', evt.loaded / evt.total * width.replace("px","")); } function getPosition(name) { var obj = document.getElementById(name); var topValue = 0, leftValue = 0; while (obj) { leftValue += obj.offsetLeft; obj = obj.offsetParent; } finalvalue = leftValue; return finalvalue; } function SetValues() { var xPos = xMousePos; var divPos = getPosition("duration_background"); var divWidth = xPos - divPos; var Totalwidth = $('#duration_background').css('width').replace("px","") audio_player.currentTime = divWidth / Totalwidth * audio_duration; $('#duration_bar').css('width', divWidth); } </script> </head> <script type="text/javascript" src="js/MousePosition.js" ></script> <body onLoad="pageLoaded();"> <table> <tr> <td valign="bottom"><input id="playButton" type="button" onClick="playClicked(this);" value=">"/></td> <td colspan="2" class="style1" valign="bottom"> <div id='player'> <div id="duration" class='player_control' > <div id="duration_background" onClick="SetValues();"> <div id="loader" style="background-color: #00FF00; width: 0px;"></div> <div id="duration_bar" class="duration_bar"></div> </div> </div> </div> </td> </tr> <tr> <td> </td> <td> <span id="currentTime">0:00</span> </td> <td align="right" > <span id="totalTime">0:00</span> </td> </tr> </table> <audio id='aplayer' src='<%=getDownloadLink() %>' type="audio/ogg; codecs=vorbis" onProgress="Progress(event);" onTimeUpdate="update();" onEnded="trackEnded();" > <b>Your browser does not support the <code>audio</code> element. </b> </audio> </body>

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  • Android RTSP coding problem

    - by NetApex
    I have Googled my butt off trying to find where if there is a surefire way to make rtsp work. I have a radio station that I listen to that streams via rtsp. Of course by default Android doesn't want to play it. If I pop the URL into yourmuze.fm and create a station there it lets me stream it to my phone. After checking how it works I come to find that it streams to the phone via rtsp! So obviously there is something amiss. What makes one stream work and one not? This is the stream I am attempting : rtsp://wms2.christiannetcast.com/yes-fm It is an audio stream so I would be thrilled with most peoples problem of "it only does audio and not video." When yourmuze.fm streams, DDMS states it brings up MovieView to play the audio if that helps at all.

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  • Embedding wav files in AS3 Flash/Flex project?

    - by aaaidan
    The Flash IDE is capable of embedding many types of uncompressed sound files, including wav, and offers optional compression when publishing. However, the [Embed] tag, only seems to allow embedding of mp3 files. Is it truly impossible to embed an uncompressed wav file, or am I missing some magic, undocumented mimeType? I was hoping for something like: [Embed source="../../audio/wibble.wav" mimeType="audio/wav"] ...but I get no transcoder registered for mimeType 'audio/wav' It's possible to embed wav or other format as an octet-stream and parse at runtime, but that's pretty heavy handed I think. I'm surprised that even though the Flash IDE can embed uncompressed sound data, [Embed] cannot, given that the swf spec can contain uncompressed sound data. Any takers?

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  • Streaming server required with JW Player?

    - by Aaron
    Currently, a site I developed plays mp3 files directly in JW Player using the file attribute and public URLs to the mp3 file. This is now an issue with the client for legal reasons, and they now need to stream the audio files so that the users can't open up their cache and nab the files directly after downloading. The JW player site has a bunch of examples for streaming video, but nothing for audio. Is it possible to stream audio files with JW player, and do we have to pay a lot of money for a streaming provider? Is it possible to do on the local php server?

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  • How to do a sample rate conversion in Windows (and OSX)

    - by Paperflyer
    I am about to write an audio file converter for my side job at the university. As part of this I would need sample rate conversion. However, my professor said that it would be pretty hard to write a sample rate converter that was both of good quality and fast. On my research on the subject, I found some functions in the OSX CoreAudio-framework, that could do a sample rate conversion (AudioConverter.h). After all, an OS has to have some facilities to do that for its own audio stack. Do you know a similar method for C/C++ and Windows, that are either part of the OS or open source? I am pretty sure that this function exists within DirectX Audio (XAudio2?), but I seem to be unable to find a reference to it in the MSDN library.

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  • Can anyone give me a sample DSP script in C/C++

    - by Andrew
    Im working on a (Audio) DSP project and just wondering if there are any sample (Open source) DSP example that are written in c or c++, for my MSP430 Chip. I just want something as a guideline so i can program my own script using the ACD and DCA on my board for sampling. http://focus.ti.com/docs/toolsw/folders/print/msp-exp430f5438.html Thats my board, MSP430F5438 Experimenter Board, from what i herd it can run dsp script via the USB connection with the computer. Im using CCS ( From TI, code composer studio) and Octave/Matlab. Just any DSP example scripts or sites that will help me create my own would be appreciated. What im tying to do, Partial audio (sampled) track -- Nyquist rate sampling -- over- and undersampling -- reconstruction of the audio track.

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  • Determining the magnitude of a certain frequency on the iPhone

    - by eagle
    I'm wondering what's the easiest/best way to determine the magnitude of a given frequency in a sound. It's my understanding that a FFT function will return the magnitudes of all frequencies in a signal. I'm wondering if there is any shortcut I could use if I'm only concerned about a specific frequency. I'll be using the iPhone mic to record the audio. My guess is that I'll be using the Audio Queue Services for recording since I don't need to record the audio to a file. I'm using SDK 4.0, so I can use any of the functions defined in the Accelerate framework (e.g. FFT functions) if needed. Update: I updated the question to be more clear as per Conrad's suggestion.

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  • feature extraction from acoustic signals

    - by Dolphin
    Hi everyone, It's been a while. I found APIs in Java for extracting features from acoustic audio files and symbolic files separately. But now I have a problem in mapping from low level wav audio features to high level midi features. i.e. I need to write the extracted wav audio features on to midi format. But I cannot think of anything even close to it. Can someone pls provide me some insight as in how I can approach this. Greatly appreciate your responses. Advance thanks

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  • How to stream a WAV file?

    - by jonasb
    I'm writing an app where I record audio and upload the audio file over the web. In order to speed up the upload I want to start uploading before I've finished recording. The file I'm creating is a WAV file. My plan was to use multiple data chunks. So instead of the normal encoding (RIFF, fmt , data) I’m using (RIFF, fmt , data, data, ..., data). The first issue is that the RIFF header wants the total length of the whole file, but that is of course not known when streaming the audio (I’m now using an arbitrary number). The other problem is that I'm not sure if it's valid since Audacity doesn't recognise the file, and Windows Media Player opens the file but plays only a very small part. I've been reading WAV specs but haven’t found an answer. Any suggestions?

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  • How to use files/streams as source/sink in PulseAudio

    - by Nilesh
    I'm a PulseAudio noob, and I'm not sure if I'm even using the correct terminology. I've seen that PulseAudio can perform echo cancellation, but it needs a source and a sink to filter from, and a new source and sink. I can provide my mic and my audio-out as the source and sink, right? Now, here's my situation: I have two video streams, say, rtmp streams, or consider two flv files, say at any given moment, stream X is the input stream that's coming from another computer's webcam+mic and stream Y is the output stream that I'm sending, (and it's coming from my computer's webcam+mic). Question: Back to the first paragraph - here's the thing, I don't want to use my mic and my audio-out, instead, I want to use these two "input" and "output" streams as my source and sink so to speak (of course, I'll use xuggler maybe, to extract just the audio from X and Y). It may be a strange question, and I have my reasons for doing this strange this - I need to experiment and verify the results to see.

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  • C# - .WAV Playback Randomly High Pitch

    - by Nate Shoffner
    For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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  • Sending series of images to display like a movie on iPhone

    - by unknownthreat
    Allow me to elaborate more. On the server, we will have a program that will take data from iPhone and process that data and produce series of images. Each time an image is generated, it will be send back to display on iPhone. I have done all of the things above using UDP, OpenGL, and such. It works. The images are transferred to iPhone and can be displayed, but it is slow. The image's resolution is around 320 x 420 and we send the image pixels by pixels. This naive implementation leads to a slow framerate. I can see around 2-3 frames per second. There are also some UDP packets dropped, and this is expected. Are there any sort of compression method available for something like this? Are there any other method that can make this better? NOTE: please don't just write "compression" as an answer, because we are aware that we will need to do it in some ways.

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  • Kde no sound from Phonon or most KDE apps but mplayer,skype and firefox are ok

    - by zeonglow
    Can somebody tell me why I cannot get any sound with most of KDE 4? I'm running a Gentoo box, I'm in both the 'audio' and 'video' groups. I can get sound with mplayer ( but not smplayer ) Firefox and Skype but nothing else. I can't get the test sound to play from the settings window, but Phonon is not whining about broken sound cards when I start up. I have checked with kmix, we seem to be completely unmuted ( and I can get sound with some apps)

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