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  • Running lame from php

    - by gok
    I am trying to run lame from a php script. I have tried these, but no luck, I don't get anything returned! Any ideas? system('lame', $returnarr); system('lame --help', $returnarr); exec('lame', $returnarr); passthru('lame', $returnarr); even this one returns nothing: exec('which lame', $returnarr); I am on OSX and final deployment will be on Linux. Do you have better suggestions for an automated wav-mp3 conversion? From php, should I execute a bash script that executes Lame?

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  • How to play sound in Python WITHOUT interrupting music/other sounds from playing

    - by Morlock
    I'm working on a timer in python which sounds a chime when the waiting time is over. I use the following code: from wave import open as wave_open from ossaudiodev import open as oss_open def _play_chime(): """ Play a sound file once. """ sound_file = wave_open('chime.wav','rb') (nc,sw,fr,nf,comptype, compname) = sound_file.getparams( ) dsp = oss_open('/dev/dsp','w') try: from ossaudiodev import AFMT_S16_NE except ImportError: if byteorder == "little": AFMT_S16_NE = ossaudiodev.AFMT_S16_LE else: AFMT_S16_NE = ossaudiodev.AFMT_S16_BE dsp.setparameters(AFMT_S16_NE, nc, fr) data = sound_file.readframes(nf) sound_file.close() dsp.write(data) dsp.close() It works pretty good, unless any other device is already outputing sound. How could I do basically the same (under linux) without having the prerequisite that no sound is being played? If you think the process would require an API to ensure software mixing, please suggest a method :) Thx for the support

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  • Manipulating multi-track ogg files programatically

    - by Chad Birch
    I'm planning to create a program for manipulating multi-track OGG files, but I don't have any experience with the relevant libraries, so I'm looking for recommendations about which language/library to use for this. I don't really have any preference for the language, I'll happily code it in C, C#, Python, whatever makes things the easiest (or even possible). Perhaps it's even a possibility to automate Audacity somehow? In terms of requirements, I'm not looking for anything particularly fancy. It will probably be a command-line program, I don't need to be able to play the audio, draw image representations of the waveforms, etc. The program will basically be used as a converter, but I need to do some processing before outputting. That is, I need the ability to programatically remove some tracks, set panning per-track, change track volumes, etc. Nothing too complex, just some basic processing, and then output the result in either MP3 or a format easily converted to MP3, such as WAV. Any suggestions or general information would be appreciated, thanks.

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  • Java sound doesn't work under Linux

    - by Cliff
    Help! I'm getting frustrated by the individual hoops I have to go through to eek sound out of my speakers when running Java apps on Linux platforms! I just installed Fedora 12 and after downloading and running the Java Sound Demo I get exceptions. If I run just a vanilla Java program that plays a wav file it runs silently with no sound and no exceptions. Every other app seems to play sound. I also took some advice from this thread in the Ubuntu forums which almost seemed to work. (Installing aoss got rid of the initial exceptions in the sound demo but I still hear nothing when I play.) can somebody help me figure out what's wrong?

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  • How to burn an Audio CD programmatically in Mac OS X

    - by Adion
    All the info I can find about burning cd's is for Windows, or about full programs to burn cd's. I would however like to be able to burn an Audio CD directly from within my program. I don't mind using Cocoa or Carbon, or if there are no API's available to do this directly, using a command-line program that can use a wav/aiff file as input would be a possibility too if it can be distributed with my application. Because it will be used to burn dj mixes to cd, it would also be great if it is possible to create different tracks without a gap between them.

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  • What event do I need to supress to stop IE from "Dinging" when I press enter in a text box?

    - by scunliffe
    On simple forms with one text box pressing enter submits the form (and this is great for easy search forms) However on a form with multiple fields, pressing Enter in an input="text" box won't do anything (e.g. submit) but in IE it "Dings" as if you have tried to delete an undeletable object. The question is... what event do I need to suppress in IE to stop this sound? e.g. if I have a username/password form, I DO want the enter key to submit the form, but I certainly don't want the "error" sound. Example site with the sound: http://www.sears.com/shc/s/StoreLocatorView?storeId=10153&catalogId=12605 Just press Enter in any of the text fields. Ding!, Ding!, Ding! Non-IE users, the sound is the: Program Events Windows Default Beep ("Windows XP Ding.wav")

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  • use more then one sound in my iphone application using cocos2d

    - by Rana
    avc = [[MPAVController sharedInstance] avController]; //avc is AVController NSString *path = [[NSBundle mainBundle] pathForResource:@"Sound" ofType:@"wav"]; id feeder = [[MPArrayQueueFeeder alloc] initWithPaths:[NSArray arrayWithObject:path]]; [avc setQueueFeeder:feeder]; [avc play:nil]; [feeder release]; NSTimer *sound = [NSTimer scheduledTimerWithTimeInterval:(9.0) target:self selector:@selector(Gamesoundplay) userInfo:nil repeats:YES]; Hear i use tow framework for playing sound. 1. MediaPlayer.framework 2. AudioToolbox.framework I success to play the background sound with repeat after lodding the game. I also want to play sound after clicking some button action without stop the previous background sound.But when i click the other button that time background sound is stop and start again after time intereval ( 9.0 sec ) which i mention for repeated the sound time line. If anyone do this work then help me to complete my application.

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  • How can I play compressed sound files in C# in a portable way?

    - by skolima
    Is there a portable, not patent-restricted way to play compressed sound files in C# / .Net? I want to play short "jingle" sounds on various events occuring in the program. System.Media.SoundPlayer can handle only WAV, but those are typically to big to embed in a downloadable apllication. MP3 is protected with patents, so even if there was a fully managed decoder/player it wouldn't be free to redistribute. The best format available would seem to be OGG Vorbis, but I had no luck getting any C# Vorbis libraries to work (I managed to extract a raw PCM with csvorbis but I don't know how to play it afterwards). I neither want to distribute any binaries with my application nor depend on P/Invoke, as the project should run at least on Windows and Linux. I'm fine with bundling .Net assemblies as long as they are license-compatible with GPL. [this question is a follow up to a mailing list discussion on mono-dev mailing list a year ago]

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  • code to play sound programmatically using a button on the iphone

    - by Brandon
    I am trying to figure out how to hook up a button to play a sound programmatically without using IB. I have the code to play the sound, but have no way of hooking the button up to play the sound? any help? here is my code that I'm using: - (void)playSound { NSString *path = [[NSBundle mainBundle] pathForResource:@"boing_1" ofType:@"wav"]; AVAudioPlayer* myAudio = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path error:NULL]]; myAudio.delegate = self; myAudio.volume = 2.0; myAudio.numberOfLoops = 1; [myAudio play]; }

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  • Problem with recording audio in Flash (Red5, ffmpeg)

    - by AT
    I'm trying to implement a small program with Flash and php that records audio and converts it to mp3. Currently I have Red5 server up and running, I can connect to it with no problems and I can publish flv recordings to the server. When I listen to the flv with Wimpy FLV player it seems to be fine. The problem comes when I'm trying to convert it with ffmpeg on the command line. I'm simply using a command ffmpeg -i but the output wav is about 50% slower than the input. When I record 10sec, the output is 15sec and pitched down. I've also tried all kinds of bitrate settings, -nv option, etc. but nothing seems to work. I have a recent version of ffmpeg that supports nellymoser format.. Don't know what to do. Anyone have any ideas?

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  • How to use infinit live streams with JAVE library? (Java, ffmpeg)

    - by Ole Jak
    So I want to use JAVE to save mp3 radio stream to my File system. I have this code for file saving but what shall I do to save a stream (stop on timer for ex) File source = new File("source.wav"); File target = new File("target.mp3"); AudioAttributes audio = new AudioAttributes(); audio.setCodec("libmp3lame"); audio.setBitRate(new Integer(128000)); audio.setChannels(new Integer(2)); audio.setSamplingRate(new Integer(44100)); EncodingAttributes attrs = new EncodingAttributes(); attrs.setFormat("mp3"); attrs.setAudioAttributes(audio); Encoder encoder = new Encoder(); encoder.encode(source, target, attrs);

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  • Novocaine - How to loop file playback? (iOS)

    - by lppier
    I'm using Novocaine by alexbw Novocaine for my audio project. I'm playing around with the example code here for file reading. The file plays back with no problem. I would like to loop this recording with the gap between the loops - any suggestion as to how I can do so? Thanks. Pier. // AUDIO FILE READING OHHH YEAHHHH // ======================================== NSArray *pathComponents = [NSArray arrayWithObjects: [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject], @"testrecording.wav", nil]; NSURL *inputFileURL = [NSURL fileURLWithPathComponents:pathComponents]; NSLog(@"URL: %@", inputFileURL); fileReader = [[AudioFileReader alloc] initWithAudioFileURL:inputFileURL samplingRate:audioManager.samplingRate numChannels:audioManager.numOutputChannels]; [fileReader play]; [fileReader setCurrentTime:0.0]; //float duration = fileReader.getDuration; [audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels) { [fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels]; NSLog(@"Time: %f", [fileReader getCurrentTime]); }];

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  • Play wave file using AudioFormat in java

    - by angelina
    Dear all, I m getting following exception while running my code on linux operating system.This code works fine on windows operating system.below is the exception and code used. java.lang.IllegalArgumentException: No line matching interface Clip supporting format PCM_SIGNED unknown sample rate, 16 bit, stereo, 4 bytes/frame, big-endian is supported. AudioFormat format = sourceaudio.getFormat(); format = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, format.getSampleRate(), format.getSampleSizeInBits() * 2, format.getChannels(), format.getFrameSize() * 2, format.getFrameRate(), true); AudioFileFormat.Type targettype = AudioFileFormat.Type.WAVE; AudioInputStream targetaudiostream = AudioSystem.getAudioInputStream(format, sourceaudio); sourceaudio.close(); targetaudiostream.close(); System.out.println("55555555"); URL url = new URL("http://localhost:8084/newvideo/PCMfile.wav"); Clip clip = AudioSystem.getClip(); AudioInputStream ais = AudioSystem.getAudioInputStream(url); clip.open(ais); System.out.println("seconds: " + (clip.getMicrosecondLength() / 1000000));

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  • Play sound in mobile browser?

    - by John
    I want to make myself a web based stop watch for training that I can use on my Blackberry mobile browser. The stopwatch should count 3 minutes, then ring a bell, wait 1 minute, then ring another bell and then repeat. My problem is I can't seem to get sound to work on my blackberry browser. I tried using <embed src="bell.wav"> which works fine in the browser of a normal computer, but it doesn't make a sound on my blackberry. Should I build this stopwatch with Javascript and HTML or should I build it with flash?

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  • How do I determine if a packet is RTP/RTCP?

    - by Chris Holmes
    I am using SharpPCap which is built on WinPCap to capture UDP traffic. My end goal is to capture the audio data from H.323 and save those phone conversations as WAV files. But first thing is first - I need to figure out what my UDP packets are crossing the NIC. SharpPCap provides a UdpPacket class that gives me access to the PayloadData of the message. But I am unsure what do with this data. It's a Byte[] array and I don't know how to go about determining if it's an RTP or RTCP packet. I've Googled this topic but there isn't much out there. Any help is appreciated.

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  • Optimize php-fpm and varnish for a powerfull server

    - by Jim
    My setup is: Intel® Core™ i7-2600 and RAM 16 GB DDR3 RAM varnish+nginx+php-fpm+apc for a not very heavy WordPress blog with W3 Total Cache and CDN My problem is that after 55 hits per second according to blitz.io varnish starts giving out timeouts. CPU usage at this time is hardly 1%. Free memory at all time remains 10GB+. I tried benchmarking php-fpm directly with result of 150hits/s without any timeouts. But after that the CPU usage goes 100% and it stops responding. Can you help me optimize it to handle more? As i understand nginx has nothing to do over here so i dont put its config. php-fpm config listen = /tmp/php5-fpm.sock listen.allowed_clients = 127.0.0.1 user = nginx group = nginx pm = dynamic pm.max_children = 150 pm.start_servers = 7 pm.min_spare_servers = 2 pm.max_spare_servers = 15 pm.max_requests = 500 slowlog = /var/log/php-fpm/www-slow.log php_admin_value[error_log] = /var/log/php-fpm/www-error.log php_admin_flag[log_errors] = on apc extension = apc.so apc.enabled=1 apc.shm_size=512MB apc.num_files_hint=0 apc.user_entries_hint=0 apc.ttl=7200 apc.use_request_time=1 apc.user_ttl=7200 apc.gc_ttl=3600 apc.cache_by_default=1 apc.filters apc.mmap_file_mask=/tmp/apc.XXXXXX apc.file_update_protection=2 apc.enable_cli=0 apc.max_file_size=1M apc.stat=1 apc.stat_ctime=0 apc.canonicalize=0 apc.write_lock=1 apc.report_autofilter=0 apc.rfc1867=0 apc.rfc1867_prefix =upload_ apc.rfc1867_name=APC_UPLOAD_PROGRESS apc.rfc1867_freq=0 apc.rfc1867_ttl=3600 apc.include_once_override=0 apc.lazy_classes=0 apc.lazy_functions=0 apc.coredump_unmap=0 apc.file_md5=0 apc.preload_path Varnish VCL backend default { .host = "127.0.0.1"; .port = "8080"; .connect_timeout = 6s; .first_byte_timeout = 6s; .between_bytes_timeout = 60s; } acl purgehosts { "localhost"; "127.0.0.1"; } # Called after a document has been successfully retrieved from the backend. sub vcl_fetch { # Uncomment to make the default cache "time to live" is 5 minutes, handy # but it may cache stale pages unless purged. (TODO) # By default Varnish will use the headers sent to it by Apache (the backend server) # to figure out the correct TTL. # WP Super Cache sends a TTL of 3 seconds, set in wp-content/cache/.htaccess set beresp.ttl = 24h; # Strip cookies for static files and set a long cache expiry time. if (req.url ~ "\.(jpg|jpeg|gif|png|ico|css|zip|tgz|gz|rar|bz2|pdf|txt|tar|wav|bmp|rtf|js|flv|swf|html|htm)$") { unset beresp.http.set-cookie; set beresp.ttl = 24h; } # If WordPress cookies found then page is not cacheable if (req.http.Cookie ~"(wp-postpass|wordpress_logged_in|comment_author_)") { # set beresp.cacheable = false;#versions less than 3 #beresp.ttl>0 is cacheable so 0 will not be cached set beresp.ttl = 0s; } else { #set beresp.cacheable = true; set beresp.ttl=24h;#cache for 24hrs } # Varnish determined the object was not cacheable #if ttl is not > 0 seconds then it is cachebale if (!beresp.ttl > 0s) { # set beresp.http.X-Cacheable = "NO:Not Cacheable"; } else if ( req.http.Cookie ~"(wp-postpass|wordpress_logged_in|comment_author_)" ) { # You don't wish to cache content for logged in users set beresp.http.X-Cacheable = "NO:Got Session"; return(hit_for_pass); #previously just pass but changed in v3+ } else if ( beresp.http.Cache-Control ~ "private") { # You are respecting the Cache-Control=private header from the backend set beresp.http.X-Cacheable = "NO:Cache-Control=private"; return(hit_for_pass); } else if ( beresp.ttl < 1s ) { # You are extending the lifetime of the object artificially set beresp.ttl = 300s; set beresp.grace = 300s; set beresp.http.X-Cacheable = "YES:Forced"; } else { # Varnish determined the object was cacheable set beresp.http.X-Cacheable = "YES"; if (beresp.status == 404 || beresp.status >= 500) { set beresp.ttl = 0s; } # Deliver the content return(deliver); } sub vcl_hash { # Each cached page has to be identified by a key that unlocks it. # Add the browser cookie only if a WordPress cookie found. if ( req.http.Cookie ~"(wp-postpass|wordpress_logged_in|comment_author_)" ) { #set req.hash += req.http.Cookie; hash_data(req.http.Cookie); } } # vcl_recv is called whenever a request is received sub vcl_recv { # remove ?ver=xxxxx strings from urls so css and js files are cached. # Watch out when upgrading WordPress, need to restart Varnish or flush cache. set req.url = regsub(req.url, "\?ver=.*$", ""); # Remove "replytocom" from requests to make caching better. set req.url = regsub(req.url, "\?replytocom=.*$", ""); remove req.http.X-Forwarded-For; set req.http.X-Forwarded-For = client.ip; # Exclude this site because it breaks if cached if ( req.http.host == "sr.ituts.gr" ) { return( pass ); } # Serve objects up to 2 minutes past their expiry if the backend is slow to respond. set req.grace = 120s; # Strip cookies for static files: if (req.url ~ "\.(jpg|jpeg|gif|png|ico|css|zip|tgz|gz|rar|bz2|pdf|txt|tar|wav|bmp|rtf|js|flv|swf|html|htm)$") { unset req.http.Cookie; return(lookup); } # Remove has_js and Google Analytics __* cookies. set req.http.Cookie = regsuball(req.http.Cookie, "(^|;\s*)(__[a-z]+|has_js)=[^;]*", ""); # Remove a ";" prefix, if present. set req.http.Cookie = regsub(req.http.Cookie, "^;\s*", ""); # Remove empty cookies. if (req.http.Cookie ~ "^\s*$") { unset req.http.Cookie; } if (req.request == "PURGE") { if (!client.ip ~ purgehosts) { error 405 "Not allowed."; } #previous version ban() was purge() ban("req.url ~ " + req.url + " && req.http.host == " + req.http.host); error 200 "Purged."; } # Pass anything other than GET and HEAD directly. if (req.request != "GET" && req.request != "HEAD") { return( pass ); } /* We only deal with GET and HEAD by default */ # remove cookies for comments cookie to make caching better. set req.http.cookie = regsub(req.http.cookie, "1231111111111111122222222333333=[^;]+(; )?", ""); # never cache the admin pages, or the server-status page, or your feed? you may want to..i don't if (req.request == "GET" && (req.url ~ "(wp-admin|bb-admin|server-status|feed)")) { return(pipe); } # don't cache authenticated sessions if (req.http.Cookie && req.http.Cookie ~ "(wordpress_|PHPSESSID)") { return(lookup); } # don't cache ajax requests if(req.http.X-Requested-With == "XMLHttpRequest" || req.url ~ "nocache" || req.url ~ "(control.php|wp-comments-post.php|wp-login.php|bb-login.php|bb-reset-password.php|register.php)") { return (pass); } return( lookup ); } Varnish Daemon options DAEMON_OPTS="-a :80 \ -T 127.0.0.1:6082 \ -f /etc/varnish/ituts.vcl \ -u varnish -g varnish \ -S /etc/varnish/secret \ -p thread_pool_add_delay=2 \ -p thread_pools=8 \ -p thread_pool_min=100 \ -p thread_pool_max=1000 \ -p session_linger=50 \ -p session_max=150000 \ -p sess_workspace=262144 \ -s malloc,5G" Im not sure where to start, should i for start optimize php-fpm and then go to varnish or php-fpm is at its max right now so i should start looking for the problem in varnish?

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  • Microsoft Speech Recognition in web service

    - by Diana
    I'm trying to use the System.Speech.Recognition namespace for recognizing speech in a web service. Actually, the WS calls a dll that uses this namespace. The problem is that...I can't. First, I had a Access denied error. After changing the Identity of my application pool to LocalSystem (security break, I know), that disappeared. But a timeout appeared. I receive no error, but no response either. I did some tests, and, the same code (very simple) that I use for recognizing the text in a WAV returns the answer in around 2 seconds, when integrated in a desktop application, but hangs and does nothing in a web application. I think I'm missing something... I'm not supposed to use System.Speech.Recognition in a web application? Am I supposed to use something else? Any help is greatly appreciated. Thank you!

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  • How to get musicbrainz track information from audio file

    - by Baki
    Can anyone tell me how to get track information from the MusicBrainz database from an audio file (mp3, wav, wma, ogg, etc...) using audio fingerprinting. I'm using MusicBrainz Sharp library, but any other library is ok. I've seen that you must use the libofa library, that you can't use MusicBrainz Sharp to get puid from the audio file, but I can't figure out how to use libofa with C#. Please show some examples and code snippets to help me, because I can't find them anywhere. Thanks in advance!

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  • Python/Tkinter Audio Player

    - by Nicholas Quirk
    Hey everyone reading this, I've recently got into doing GUI development with Python. Tkinter seems like the easiest and most logical choice starting out. I did a little with wxPython but it was more sophisticated than what I needed. Anyway, I'm developing a media player. Right now it's a simple window with a button to load .wav files. The problem is I would like to implement a pause button now. But, when playing a audio file the GUI isn't accessible again (no buttons can be pushed) till the file is done playing. How can I make the GUI dynamic while an audio file is playing? I was thinking this maybe be because I'm using PyAudio, and their implementation doesn't allow this. Anyway, thanks for any advice before hand.

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  • Client-side framework for web-app with good audio support

    - by Poita_
    I'm trying to create a client-side web app that generates music procedurally using some user-input parameters, so I'm looking for a framework (e.g. Flash, Silverlight etc.) that has the capability to play audio at a specified pitch. Whether it is playing a WAV/MP3 file, using MIDI output, or just playing beeps doesn't really matter -- I just need something that will enable me to generate arbitrary music client-side. I've done a bit of searching and it appears that Flash might have the ability to change pitch with the help of a third-part plugin, but I couldn't find anything similar for Silverlight. I can go a try all them out manually if need be, but I thought I'd ask here first just in case anyone had tried something like this before. Thanks in advance

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  • Vista not supporting trueSpeech!

    - by csmba
    My web application has (server side) a wav file saved using truespeech and then a client (ie6, ie7) can ask to plat the file and the web server serves it.. All you need on xp is to have WMP 9 or higher and it all works. but on vista.. suddenly the vista client box can't play the file cause it doesn't support truespeech (some upgrade!) anyone has an idea of what should I do? You can suggest a way to make it work on the client (but in general, I don't like having to say that the solution includes installing anything on the client box) you can suggest I don't save the server side file in truespeech, and instead use something else (what?)

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  • Audio Detection in Matlab

    - by insane-36
    I am writing a matlab code that would be able to read the audio file and then compare it to the another audio and recognize if those audio are the voice of the same person. In both type of the audio, would have the same word utterance and the audio is about 1 minutes long. I have come to know that the approach of sliding windows using hamming window would work best on this approach but have a very little idea on this. The simple code to read an audio file and then display a portion of 10s is as below : [x,fs, nbits]= wavread('01-AudioTrack 01.wav'); subplot(211) plot(x) title('Entire Wave') smallRange = 1:100000; subplot(212) plot(smallRange,x(smallRange)) How do I make Hamming window each of 10ms in this case and what approaches should I take to deal with this problem ?

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  • set different volumes inside app

    - by blacksheep
    i'd lioke to set the volume inside the touchesBegan action on half the volume of the IBAction. (void) awakeFromNib { [super awakeFromNib]; engine = [[Finch alloc] init]; E = [[RevolverSound alloc] initWithFile:PATH(@"E.wav")rounds:9]; } -(void)touchesBegan:(NSSet *)touches withEvent:(UIEvent *)event { UITouch *touch = [[event allTouches] anyObject]; CGPoint location = [touch locationInView:self.view]; if(CGRectContainsPoint(Edrop.frame,location)){ [E play]; } } (IBAction)bass:(id)sender { if(CGRectIntersectsRect(finga.frame,e.frame)){ if(finga.center.y <= e.center.y) [E play]; } } thanx, blacksheep

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  • GUI Control For Audio Presentation

    - by Boris
    I need GUI control for audio file presentation. The language is not very important but it should run on windows platform. I should be able to :- load the file play the sound put and move markers across the audio bar. it would be nice if it can load itself from RTP wireshark captures (and not wav files). An example may be seen in audacity (may be someone even had an experience extracting it from there). Writing nyquist scripts in audacity is not a good option because I have to operate on RTP captures and not on raw sound samples. Another example of such control is wireshark RTP analyzer. Any advise?

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  • Anyone know of a .net library/utility that will convert a word document to an mp3 format

    - by EJB
    Anyone know of any well-supported/proven methods for converting a Microsoft word document to an MP3 or wav format such that hearing-impaired folks could "listen" to documents that I have stored in my web-based document management system? I already have the interface built such that someone can use the telephone to get the list of documents available, with the dates and titles "read" to them over the phone, but now I would like the ability to let someone actually listen to the contents of word files stored in the system. Ideally a .net library or utility that would let me convert the DOC - MP3 after each upload would be best, but one that "read" the file on demand would be OK too.

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