I've noticed that if my audio files are in caf 11250Hz mono they perform worse than 44.1Khz mono. Tracing it with profiler I can see that for the low sample rate files one of the longest traces ends with LinearConverterInt32. This isn't present in the 44.1KHz trace.
I want to use the lower sample rate files to keep file size (and hopefully memory size) down.
I've noticed in my log file that I get this AudioStreamBasicDescription: 2 ch, 44100 Hz, 'lpcm' (0x00000C2C) 8.24-bit little-endian signed integer, deinterleaved
So I am guessing that this is the format it is converting to, but I have no idea how to tell it to use 1 ch, 11250 Hz 16 bit.
Thoughts?