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• Nyquist won't play audio

  - by erjiang

  I downloaded Nyquist, and am having trouble playing sounds from it. If I run it normally, I get: Nyquist -- A Language for Sound Synthesis and Composition Copyright (c) 1991,1992,1995 by Roger B. Dannenberg Version 2.29 > (play (osc 60)) Saving sound file to ./eric-temp.wav error: snd_save -- could not open audio output > If I wrap it by running padsp ny, the sound plays fine for about half a second, and then I get garbage fed to my speakers. Any solutions?

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• Does this schema sound better suited for a document-oriented data store or relational?

  - by Blaine LaFreniere

  Disclaimer: let me know if this question is better suited for serverfault.com I want to store information on music, specifically: genres artists albums songs This information will be used in a web application, and I want people to be able to see all of the songs associated to an album, and albums associated to an artist, and artists associated to a genre. I'm currently using MySQL, but before I make a decision to switch I want to know: How easy is scaling horizontally? Is it easier to manage than an SQL based solution? Would the above data I want to store be too hard to do schema-free? When I think association, I immediately think RDBMSs; can data be stored in something like CouchDB but still have some kind of association as stated above?

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• Problem with playing a sound when a button is clicked.

  - by iSharreth

  in my code: import "MyViewController.h" import (IBAction)playSound{ AVAudioPlayer *myExampleSound; NSString *myExamplePath = [[NSBundle mainBundle] pathForResource:@"myaudiofile" ofType:@"caf"]; myExampleSound =[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:myExamplePath] error:NULL]; myExampleSound.delegate = self; [myExampleSound play]; } But it is showing a warning that Class MyViewController does not implement AVAudioplayerDelegate. Anyone please help. I had included the AVFoundation.Framework.

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• What do you use to play sound in iPhone games?

  - by zoul

  Hello! I have a performance-intensive iPhone game I would like to add sounds to. There seem to be about three main choices: (1) AVAudioPlayer, (2) Audio Queues and (3) OpenAL. I’d hate to write pages of low-level code just to play a sample, so that I would like to use AVAudioPlayer. The problem is that it seems to kill the performace – I’ve done a simple measuring using CFAbsoluteTimeGetCurrent and the play message seems to take somewhere from 9 to 30 ms to finish. That’s quite miserable, considering that 25 ms == 40 fps. Of course there is the prepareToPlay method that should speed things up. That’s why I wrote a simple class that keeps several AVAudioPlayers at its disposal, prepares them beforehand and then plays the sample using the prepared player. No cigar, still it takes the ~20 ms I mentioned above. Such performance is unusable for games, so what do you use to play sounds with a decent performance on iPhone? Am I doing something wrong with the AVAudioPlayer? Do you play sounds with Audio Queues? (I’ve written something akin to AVAudioPlayer before 2.2 came out and I would love to spare that experience.) Do you use OpenAL? If yes, is there a simple way to play sounds with OpenAL, or do you have to write pages of code? Update: Yes, playing sounds with OpenAL is fairly simple.

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• Is it possible to detect when the system is recording a sound and then perform some action on Python

  - by Jorge

  I began learning Python a few days ago, and i was wondering about a practical use for a program. Then i came up with the following: if my brother is in his room recording himself playing guitar, a led plugged to the usb and wired so it's outside his door lights up, and then i'll know he's recording and i'll take care not to make any noises. The main questions are: How Python can detect any recording going on in the system? How would i interface with the usb so i can actually turn the led on?

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• Audio broken on clean install of ThinkPad T60

  - by Ben Alpert

  I just reinstalled a clean version of XP SP3 on a ThinkPad T60. After downloading all of the missing drivers from the Lenovo site, everything seems to work except for the audio. I installed the audio driver from http://www-307.ibm.com/pc/support/site.wss/MIGR-62928.html (version 5.10.1.4326) so SoundMAX shows up in the control panel and everything in the Sound preferences looks okay, but no sound comes out of the speakers when I try to play music or hear the interface sound effects. There's also no sound if I plug in headphones and listen. Some forums suggested installing hotfix Q888111 but it's already included in SP3 so I can't install it. I also tried reinstalling the audio driver mentioned above in case something went wrong the first time, but it still doesn't work. Does anyone know how to make the sound work?

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• upgrade:upgradation failed,13.10 issue

  - by Atul

  I was upgrading 13.04 to 13.10.And suddenly it hanged and upper part of the screen went black. So I stopped the upgradation and unfortunately I forcefully shutdown.But its already upgraded to 13.10(I feel upgradation is not complete though) Problems The desktop flickers now and sometimes the screen turns black. Sound stopped.No sound output detected in sound settings.And the shutdown button also not working. Is there a way to upgrade it again to 13.10(completely)?

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• Headphones don't work on an Acer Aspire 5920g

  - by Dave Chatelain

  All the problems I read about were all about the sound not working when they plugged the headphones in. I have the problem that my mic will work but the sound is still played on my built-in speakers. When I look at the sound settings, it doesn't give me any options for headphones. I run 12.04 and never had this problem in Windows. I run it on a Acer Aspire 5920g. Any tips or ideas will be very helpful.

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• Visually and audibly unambiguous subset of the Latin alphabet?

  - by elliot42

  Imagine you give someone a card with the code "5SBDO0" on it. In some fonts, the letter "S" is difficult to visually distinguish from the number five, (as with number zero and letter "O"). Reading the code out loud, it might be difficult to distinguish "B" from "D", necessitating saying "B as in boy," "D as in dog," or using a "phonetic alphabet" instead. What's the biggest subset of letters and numbers that will, in most cases, both look unambiguous visually and sound unambiguous when read aloud? Background: We want to generate a short string that can encode as many values as possible while still being easy to communicate. Imagine you have a 6-character string, "123456". In base 10 this can encode 10^6 values. In hex "1B23DF" you can encode 16^6 values in the same number of characters, but this can sound ambiguous when read aloud. ("B" vs. "D") Likewise for any string of N characters, you get (size of alphabet)^N values. The string is limited to a length of about six characters, due to wanting to fit easily within the capacity of human working memory capacity. Thus to find the max number of values we can encode, we need to find that largest unambiguous set of letters/numbers. There's no reason we can't consider the letters G-Z, and some common punctuation, but I don't want to have to go manually pairwise compare "does G sound like A?", "does G sound like B?", "does G sound like C" myself. As we know this would be O(n^2) linguistic work to do =)...

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• Microphone not working at all on a Toshiba sattelite p505 s8980

  - by flamingburner

  My built-in microphone doesn't work in "record audio" program and in Skype. I'm using ubuntu 11.10 in Toshiba sattelite p505 s8980. No mute button checked I have PulseAudio Volume Control installed currently when aim trying to use record audio program and hear what I record it plays noisy sound not my voice I changed my computer from 'Analog Stereo Output' to 'Analog Stereo Duplex', in both sound setting ( ubuntu ) and in PulseAudio Volume Control no problem in bios and in my windows seven sound worked perfectly and in ubuntu no problem in sound at all i can play music and movies and even in skype i can hear the sounf of who is taking to me ( it's about the microphone only "

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• send music to upnp device from PC

  - by markrich

  I have a new Arcam AirDAC (http://www.arcam.co.uk/products,rSer...ACs,airDAC.htm) attached to my stereo which has upnp support. I would like to send audio from my 14.04 PC to the box itself and to this end installed Rygel upon my system to help but it hasn't. I have created a new sound device in PulseAudioPreferences and selected it from inidcator-sound-switcher but here I become stuck. The sound is heading to the new sound device as the volume can be seen to go up and down from PulseAudioVolumeControl but no sound comes from the stereo downstairs. The problem, as I see it, is the new device has no idea where to send the music as the Arcam hasn't been chosen from any program. So - I installed BubbleUPNP and Plex. My music has been imported into the later and the former can see both the Arcam as a Renderer and the Plex as the Media Server. Installing the BubbleUPNP program on my Android tablet allowed me to send music and all seemed good UNTIL I started playing AIFF and ALAC music and it all stopped. No suitable decoding device. So that scuppered that route. So here I am and stuck. How can I tell Ubuntu to use the Arcam as a renderer to play music through when the albums are played from Rhytmnbox, Tomahawk, Clementine or other? Clementine would be my preferred client as there is a usable remote control program for the tablet. Can anyone help me fix this or advice another way to do what I would like?

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• iphone, how to play sound even in silent or mute mode ?

  - by Robin

  as topic... is it possible ? Thanks

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• ASUS Xonar DG not working Ubuntu

  - by thephpdev

  I've checked the "Questions that may already have your answer" to no avail. Basically, there is no sound at all, the snd_oxygen drivers are all being loaded. I can see the sound card and all of it's separate ports in Settings-Sound-Output, but nothing is coming out. I tried adjusting the volume in alsamixer, but for some reason the volume won't change (all that is there are some options [Speakers, Headphones, FP Headphones] and they change when I press the up/down arrow keys) at all. I have also tried many other terminal commands, such as reloading alsa and checking if the sound card is even being detected, which it is. I'm left clueless. It says on a bug report that this has been unassigned for over a year. Surely that's not right. Thanks in advance, Caelan.

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• sometimes, strange audio distortion(peeping, scratching), ubuntu 12.04

  - by richi902

  i have a problem with my sound in ubuntu 12.04. the problem is, that sometimes out of nowhere, when switching songs, or playing youtube videos, changing volume with my keyboard buttons, that the sound gets distorted(peeping, scratching). i dont know if it is related, but when i skip through music in rythmbox, there is also a little scratching noise. i can sometimes temporarly fix it: for youtube videos, i refresh the page, and sometimes it works agian normal, mostly not. for audio playback with rythmbox, i have to pause the song for sometime, and resume it, and hope that it works. before all that,i have changed my soundcard to "Analog Surround 5.1" in the sound settings from ubuntu, but i also used alsamixer to change it from 2 channels to 6 channels, since changing in ubuntu sound settings alone wasnt enough to make the other speaks work. i use a ASUS P8-H61-M LE B3 Revision Motherboard. which has a built in surround soundcard.

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• Is there a way to programmatically extract the sound portion of a Flash SWF file with .NET?

  - by hypoxide

  Do any libraries exist with this functionality? Any recommendations would be useful. Thanks.

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• Visual Studio IDE: I want it to make a sound after it compiles so I can get back to work.

  - by user341550

  Would be a nice thing - do I have to write a script or can't I just tweak the build script to do some task?

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• Chrome will not load a web page with an <embed> element

  - by rossmcm

  I have been trying to get a simple sound web page going: Sound.html <script> function PlaySound () { var sounder = document.getElementById ("ToneA") ; sounder.Play () ; } </script> <embed id="ToneA" height="1" width="1" src="https://dl.dropboxusercontent.com/u/311035/ToneA.mp3" autostart="false" enablejavascript="true"//> <button onclick="PlaySound () ;">Play</button> The test web page is here. It plays in IE, but not in Firefox or Chrome. My problem: Chrome reports "Could not load VLC Plugin". It seems to be a known problem that the VLC community don't necessarily feel like fixing at the moment, and is a result of Google choosing not to allow some certain kind of plugin. If I disable the plugin I no longer get the message but nothing happens when I click the button. Looking at the console in a debug window I see Uncaught TypeError: undefined is not a function Sound.html:7 PlaySound Sound.html:7 onclick which suggests Chrome could not find anything else to handle the sound file. How to I tell Chrome to use (e.g.) Windows Media Player? * UPDATE * This is apparently because the VLC plugin is a NPAPI plugin and Chrome no longer supports these. I have uninstalled VLC and this has removed the error on loading the webpage with an embedded sound element, but it still doesn't invoke WMP instead.

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• Objective-c - How to serialize audio file into small packets that can be played?

  - by vfn

  Hi there, So, I would like to get a sound file and convert it in packets, and send it to another computer. I would like that the other computer be able to play the packets as they arrive. I am using AVAudioPlayer to try to play this packets, but I couldn't find a proper way to serialize the data on the peer1 that the peer2 can play. The scenario is, peer1 has a audio file, split the audio file in many small packets, put them on a NSData and send them to peer2. Peer 2 receive the packets and play one by one, as they arrive. Does anyone have know how to do this? or even if it is possible? EDIT: Here it is some piece of code to illustrate what I would like to achieve. // This code is part of the peer1, the one who sends the data - (void)sendData { int packetId = 0; NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:@"myAudioFile" ofType:@"wav"]; NSData *soundData = [[NSData alloc] initWithContentsOfFile:soundFilePath]; NSMutableArray *arraySoundData = [[NSMutableArray alloc] init]; // Spliting the audio in 2 pieces // This is only an illustration // The idea is to split the data into multiple pieces // dependin on the size of the file to be sent NSRange soundRange; soundRange.length = [soundData length]/2; soundRange.location = 0; [arraySoundData addObject:[soundData subdataWithRange:soundRange]]; soundRange.length = [soundData length]/2; soundRange.location = [soundData length]/2; [arraySoundData addObject:[soundData subdataWithRange:soundRange]]; for (int i=0; i // This is the code on peer2 that would receive an play the piece of audio on each packet - (void) receiveData:(NSData *)data { NSKeyedUnarchiver* unarchiver = [[NSKeyedUnarchiver alloc] initForReadingWithData:data]; if ([unarchiver containsValueForKey:PACKET_ID]) NSLog(@"DECODED PACKET_ID: %i", [unarchiver decodeIntForKey:PACKET_ID]); if ([unarchiver containsValueForKey:PACKET_SOUND_DATA]) { NSLog(@"DECODED sound"); NSData *sound = (NSData *)[unarchiver decodeObjectForKey:PACKET_SOUND_DATA]; if (sound == nil) { NSLog(@"sound is nil!"); } else { NSLog(@"sound is not nil!"); AVAudioPlayer *audioPlayer = [AVAudioPlayer alloc]; if ([audioPlayer initWithData:sound error:nil]) { [audioPlayer prepareToPlay]; [audioPlayer play]; } else { [audioPlayer release]; NSLog(@"Player couldn't load data"); } } } [unarchiver release]; } So, here is what I am trying to achieve...so, what I really need to know is how to create the packets, so peer2 can play the audio. It would be a kind of streaming. Yes, for now I am not worried about the order that the packet are received or played...I only need to get the sound sliced and them be able to play each piece, each slice, without need to wait for the whole file be received by peer2. Thanks!

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• Optical SPDIF audio from motherboard not working with receiver

  - by simon b

  Hi, I hope someone can help; I can't get my SPDIF optical out working through my receiver and all the responses I can see on the web assume you have a sound card, while I settled for the (seemingly high end) sound on my motherboard (Asus P7P55D-E PRO), which appears to limit some of my options. My set-up is a "new out of the box" one and is: *Windows 7 PC (using PowerDVD10 for DVDs/Blurays and Windows media player for music) *Asus P7P55D-E PRO motherboard - has 8-channel audio TRS jacks and SPDIF optical and coaxial out *An old Yamaha receiver, whose only multi-channel input options are optical in and 6 channel RCA in. However, it still can handle DTS and DD *Boston Acoustic Soundware XS 5.1 speakers I've currently got the SPDIF optical out from the motherboard connected to the in on my receiver, have SPDIF enabled in the sound menu and the light is glowing red down the fibre. But I'm getting no sound at all. What I want is to be able to play DVDs/BluRays in 5.1 but also to be able to play music in multi-channel mode (even though I know this will be "fake" multichannel; it's more about where I sit in the room and my requirement to use the sub because the Boston is a satellite/sub set-up) My questions are: *Will optical work at all for multi-channel? THe latest posts I can see suggest it does but some people seem to say optical only outputs stereo. Whom to believe? *Even if it does work, I've read that I have to disable AC-3 decoding, or make various other changes, which don't seem to be possible without the menu options that a sound-card brings. Is the motherboard-only option just too inflexible? *Although my SPDIF device is enabled in the sound menu, it insists under "Jack information" that it is a "rear panel RCA jack", when of course it is not (both TOSLINK and rCA jacks do exist). Has the PC just forgotten that it has an optical? *I think I could relatively easily connect the 8-channel 3.5mm TRS jacks to my receiver 6-ch input jacks by way of TRS/RCA cables, but would that not stop me from being able to play music from media-player in multi-channel mode, as I'm not sure the motherboard can cope *Or do I need to bite the bullet and buy a sound-card? And if so, how can I be sure the one I get doesn't have the same problem? Any thoughts gratefully received, Cheers, simon

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• How would make this run with an if statement and one for loop?

  - by Nick Jacobs

  I'm trying to get this to run by using an if statment, a for loop, and a list. The list is part of the parameters. I am not sure how to write the if statement and have the program loop through all of the different words and set everything how it is supposed to be. newSndIdx=0; for i in range (8700, 12600+1): sampleValue=getSampleValueAt(sound, i) setSampleValueAt(newSnd, newSndIdx, sampleValue) newSndIdx +=1 newSndIdx=newSndIdx+500 for i in range (15700, 17600+1): sampleValue=getSampleValueAt(sound, i) setSampleValueAt(newSnd, newSndIdx, sampleValue) newSndIdx +=1 newSndIdx=newSndIdx+500 for i in range (18750, 22350+1): sampleValue=getSampleValueAt(sound, i) setSampleValueAt(newSnd, newSndIdx, sampleValue) newSndIdx +=1 newSndIdx=newSndIdx+500 for i in range (23700, 27250+1): sampleValue=getSampleValueAt(sound, i) setSampleValueAt(newSnd, newSndIdx, sampleValue) newSndIdx +=1 newSndIdx=newSndIdx+500 for i in range (106950, 115300+1): sampleValue=getSampleValueAt(sound, i) setSampleValueAt(newSnd, newSndIdx, sampleValue) newSndIdx+=1

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• Can I play any Buffer only once at a given time?

  - by mystify

  From the OpenAL documentation: The basic OpenAL objects are a Listener, a Source, and a Buffer. There can be a large number of Buffers, which contain audio data. Each buffer can be attached to one or more Sources My problem is, that I have one sound file which I need to play multiple times per second, at the same time. The sound is 2 seconds long. So it will overlap. Would I need multiple filled buffers for this (= multiple times that sound in memory)? If I would attach one Buffer to multiple Sources, would I be able to play the sound 10 times, overlapping itself, with just one copy in memory? Or would I still have to deal with 10 copies of that sound in memory?

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• Delphi - Proper way to page though data.

  - by Brad

  I have a string list (TStrings) that has a couple thousand items in it. I need to process them in groups of 100. I basically want to know what the best way to do the loop is in Delphi. I'm hitting a brick wall when I'm trying to figure it out. Thanks unit Unit2; interface uses Windows, Messages, SysUtils, Variants, Classes, Graphics, Controls, Forms, Dialogs, StdCtrls; type TForm2 = class(TForm) Memo1: TMemo; Memo2: TMemo; Button1: TButton; procedure Button1Click(Sender: TObject); private { Private declarations } public { Public declarations } end; var Form2: TForm2; implementation Uses math; {$R *.dfm} procedure TForm2.Button1Click(Sender: TObject); var I:Integer; pages:Integer; str:string; begin pages:= ceil(memo1.Lines.Count/100) ; memo2.Lines.add('Total Pages: '+inttostr(pages)); memo2.Lines.add('Total Items: '+inttostr(memo1.Lines.Count)); // Should just do in batches of 100 VS entire list for I := 0 to memo1.lines.Count - 1 do begin if str '' then str:= str+#10+ memo1.Lines.Strings[i] else str:= memo1.Lines.Strings[i]; end; //I need to stop here every 100 items, then process the items. memo2.Lines.Add(str); end; end. Example form object Form2: TForm2 Left = 0 Top = 0 Caption = 'Form2' ClientHeight = 245 ClientWidth = 527 Color = clBtnFace Font.Charset = DEFAULT_CHARSET Font.Color = clWindowText Font.Height = -11 Font.Name = 'Tahoma' Font.Style = [] OldCreateOrder = False PixelsPerInch = 96 TextHeight = 13 object Memo1: TMemo Left = 16 Top = 8 Width = 209 Height = 175 Lines.Strings = ( '4xlt columbia thunder storm jacket' '5 things about thunder storms' 'a thunder storm with a lot of thunder ' 'and lighting sccreensaver' 'a thunder storm with a lot of thunder ' 'and lighting screensaver with no nag ' 'screens' 'all about thunder storms' 'all about thunderstorms for kids' 'amazing tornado videos and ' 'thunderstorm videos' 'are thunder storms louder in ohio?' 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'thunderstorm warning definition' 'thunderstorm warning los angeles' 'thunderstorm warning san diego' 'thunderstorm warning san mateo county' 'thunderstorm warning santa barbara' 'thunderstorm warning santa clara' 'thunderstorm warning santa clara ' 'county' 'thunderstorm warning signal' 'thunderstorm warning signs' 'thunderstorm warning vs watch' 'thunderstorm warnings' 'thunderstorm warnings and watches' 'thunderstorm warnings for nj' 'thunderstorm warnings qld' 'thunderstorms' 'thunderstorms facts' 'thunderstorms facts for kids' 'thunderstorms for kids' 'tornados and thunder storms animated' 'understanding thunderstorms for kids' 'watch thunderstorm videos' 'weather underground forecast ' 'thunderstorms' 'what causes thunder storms' 'what is a thunder storm' 'where d thunder storms occur') TabOrder = 0 end object Memo2: TMemo Left = 240 Top = 8 Width = 265 Height = 129 Lines.Strings = ( 'Memo2') TabOrder = 1 end object Button1: TButton Left = 384 Top = 184 Width = 75 Height = 25 Caption = 'Button1' TabOrder = 2 OnClick = Button1Click end end

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• why AVAudioplayer doesn't stop/pause when viewWillDisappear?

  - by Rahul Vyas

  I am using avaudioplayer in my app. on viewwilldisappear i want to pause/stop the sound and on viewwill appear i want to play sound again.how do i do this? i'm using this code on viewWillAppear:- if(self.TickPlayer) { [self.TickPlayer play]; } if(self.TickPlayer.volume<0) { self.TickPlayer.volume=1.0; } and this on viewWillDisAppear if(self.TickPlayer) { [self.TickPlayer stop]; } here is the method which plays the sound -(void)PlayTickTickSound:(NSString*)SoundFileName { //Get the filename of the sound file: NSString *path = [NSString stringWithFormat:@"%@%@",[[NSBundle mainBundle] resourcePath],[NSString stringWithFormat:@"/%@",SoundFileName]]; //Get a URL for the sound file NSURL *filePath = [NSURL fileURLWithPath:path isDirectory:NO]; NSError *error; if(self.TickPlayer==nil) { self.TickPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:filePath error:&error]; // handle errors here. self.TickPlayer.delegate=self; [self.TickPlayer setNumberOfLoops:-1]; // repeat forever [self.TickPlayer play]; } if(self.TickPlayer) { [self.TickPlayer play]; } } and i'm calling it in method which fires every second using a timer in viewDidLoad- self.timer = [NSTimer scheduledTimerWithTimeInterval:1.0 target:self selector:@selector(updateCountdown) userInfo:nil repeats:YES]; -(void) updateCountdown { [self PlayTickTickSound:@"tick.caf"]; } i'm also playing a beep sound using avaudioplayer at a specific condition when alert appears.that works fine but this is not working as expected

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• Stop all sounds permanently in AS3

  - by Fahim Akhter

  Hi, I have a main swf which has sound on/off buttons. It has many SWF's which are loaded into different placeholders at different time. All of them have different sounds in them. In addition to that there is a music loop going on in the background. Now, this perticular swf lets call it Father will be placed in some swf later on. What I'm trying to do right now is when the sound off button is pressed turn off all the sounds permanently so the child swf's and their sounds stop too. I have found a way in AS2 : global = new Sound( ) //no movie clip target path. and add the following code to your off button. on(release){ global.setVolume(0); //mutes all sound } and for on button on(release){ global.setVolume(100); //un mutes all sound } which ofcourse does not work for AS3. So how can I stop all sounds permanently in AS3? Secondly what if I have one sound (my background loop) which I want to keep going on. Would be a lot of help, been searching for a while now without a appropriate answer.

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• AVAudioPlayer not unloading cached memory after each new allocation

  - by Rob

  I am seeing in Instruments that when I play a sound via the standard "AddMusic" example method that Apple provides, it allocates 32kb of memory via the prepareToPlay call (which references the AudioToolBox framework's Cache_DataSource::ReadBytes function) each time a new player is allocated (i.e. each time a different sound is played). However, that cached data never gets released. This obviously poses a huge problem if it doesn't get released and you have a lot of sound files to play, since it tends to keep allocating memory and eventually crashes if you have enough unique sound files (which I unfortunately do). Have any of you run across this or what am I doing wrong in my code? I've had this issue for a while now and it's really bugging me since my code is verbatim of what Apple's is (I think). How I call the function: - (void)playOnce:(NSString *)aSound { // Gets the file system path to the sound to play. NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; // Converts the sound's file path to an NSURL object NSURL *soundURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; self.soundFileURL = soundURL; [soundURL release]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL: soundFileURL error:nil]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely // this is where the prior cached data never gets released [theAudio prepareToPlay]; // set it up and play [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio setDelegate: self]; [theAudio play]; } and then theAudio gets released in the dealloc method of course.

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