I'm looking to create an app that emulates a physical instrument. I've got audio samples but I want to be able to increase the pitch/frequency dynamically so I don't have to load from too many files.
Any idea which audio API will be able to do this? I reckon either OpenAL or Audio Queue Services but am not sure which is suitable. Any links to guides/sample code is also much appreciated.
Thanks in advance.
I went down this road in 2009, trying Audio Toolbox, Audio Queue Services, openAL, and finally settling on the RemoteIO AudioUnit.
Audio Toolbox is fine for basic triggered sound effects, but it wasn't able to change frequencies or loop samples.
Audio Queue Services can loop samples, but the only way I could find to adjust the playback frequency of a sample was to re-read the data from the file -- very painful. Plus, the framework is tremendously cumbersome - I'd only use it if I was trying to stream something off the Internet.
OpenAL was a godsend - was up and running with it in under an hour, after getting my hands on the no-longer-available-from-Apple "CrashLanding" iPhone sample app. I found OpenAL to be ideally suited to games or even a musical instrument -- samples could be pre-loaded, adjusting the frequency was easy, and looping was no problem. The deal-breaker for me was that starting and stopping a looped sample would result in a nasty "pop" almost every time. Also the builtin 3d positional audio mixer was a bit too CPU-intensive for my liking.
If your instrument does not use looped samples, I'd suggest trying the OpenAL route first - the learning curve is much less intimidating. Try to track down "SoundEngine.h", "CrashLanding" or "TouchFighter", or check out the following link:
http://benbritten.com/blog/2008/11/06/openal-sound-on-the-iphone/
Since looped samples was a requirement for me, I finally settled on AudioUnits (which, on the iPhone, is referred to as "RemoteIO" if you want to do input or output). It was tremendously difficult to implement - very similar to Audio Queue Services, in that the core of your implementation will be inside a "buffer callback", being called several times per second to fill a buffer of outbound audio with raw SInt16 values.
Ultimately, I got my instrument working beautifully with multi-note polyphony, looped samples, no popping, and minimal latency.
Unfortunately, RemoteIO is not well documented. Michael Tyson was one of the first in the field to write about RemoteIO at length, and his posts (and the comments) were very useful to me:
http://michael.tyson.id.au/2008/11/04/using-remoteio-audio-unit/
Good luck!
Edited years later: I've open-sourced the RemoteIO/AudioUnits code I alluded to above: https://github.com/glenn-barnett/hexaphone/blob/master/Classes/Instrument.m - apologies for the mess, I hope to get some time to clean up the code and comments.