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    <p>1) the os x/iphone file read routines allow you to determine the sample format, typically one of SInt8, SInt16, SInt32, Float32, Float64, or contiguous 24 bit signed int for LPCM</p> <p>2) for int formats, MIN_FOR_TYPE represents the max amplitude in the negative phase, and MAX_FOR_TYPE represents the maximum amplitude in the positive. 0 equals silence. floating point formats modulate between [-1...1], with zero as with float. when reading, writing, recording, or working with a specific format, endianness will matter - a file may require a specific format, and you typically want to manipulate the data in the native endianness. some routines in the apple audio file libs allow you to pass a flag denoting source endianness, rather than you manually converting it. CAF is a bit more complicated - it acts like a meta wrapper for one or more audio files, and supports many types.</p> <p>3) the amplitude representation for lpcm is just a brute-force linear amplitude representation (no conversion/decoding is required to playback, and the amplitude steps are equal).</p> <p>4) see #2. the values are not related to air pressure, they are related to 0 dBFS; e.g. if you're outputting the stream straight to a DAC, then the int max (or -1/1 if floating point) represents the level at which an individual sample will clip.</p> <p>Bonus) it, like every ADC and component chain has limits to what it can handle on input in terms of voltage. additionally, the sampling rate defines the highest frequency that may be captured (the highest being half of the sampling rate). the adc may use a fixed or selectable bit depth, but the max input voltage does not generally change when choosing another bit depth.</p> <p>one mistake you're making at the code level: you're manipulating `outBuffer' as chars - not SInt16</p>
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