I'm not using wav files "still" but again, ever since making my own DirectSound implementation. I don't want to use media_play so I'd have to decode mp3 myself before I could feed it into the sound buffers.

Ogg files are (/ can be) smaller when you use compression. Compressed audio is lossy audio, thus having worse quality (although on high quality settings rather unnoticeable). However, I'd have to use an ogg decoding library as well for that. I looked into it for a bit but didn't feel like implementing it.

Although frequently used, FMOD has a ton of issues (and used to have bad licensing for indies). Some projects are stuck with audio issues for months because of FMOD.

Previously I had used irrKlang (65 bucks license) which worked fine with automatic streaming/ memory management. There was a bug with the Windows/ DirectSound implementation though, messing sounds up when you played more than one instance of them at the same time. I had reported it and worked with the dev but he took over half a year or so to fix it, so I just wrote my own "sound engine".


It's all working fine, I just need to know that lite-C's malloc is threadsafe.