I was hoping someone could help me work out a more detailed description of some of the fundamental AudioStream elements. I am particularly interested in learning more than the given info on the "Creating New Audio Objects" page. I'd love to hear a deeper "how to use" description as well as a more technical explanation of the inner workings for a few of the basics such as audio_block_t, recieveReadonly()/recieveWritable() , transmit() and inputQueue().

Specific Questions:
I'm having trouble wrapping my head around what this means from the audio_block_t description:

"The only member intended for use in update() is "data", an array of 16 bit integers representing the audio ... The data[] array is always 32 bit aligned in memory, so you can fetch pairs of samples by type casting the address as a pointer to 32 bit data"

I am assuming that the 32 bit type cast is for getting left and right channel simultaneously, but I would love to know the logistics of how to grab a pair of samples with type casting as 32 bits as mentioned in that description.

Also where might I find the audio_block_t struct definition to look at? In AudioStream.h I found a struct called audio_block_struct but no audio_block_t definition.
typedef struct audio_block_struct {
	uint8_t  ref_count;
	uint8_t  reserved1;
	uint16_t memory_pool_index;
	int16_t  data[AUDIO_BLOCK_SAMPLES];
} audio_block_t;

A few more questions:

1. How often is update() called for an audio object? Is it called as a timer ISR after sampling 128 (or whatever the block size) number of times?
2. Does the sgtl5000 input signal full voltage range map to a block sample value of 0-65535 ?
3. Is the input queue filling up with new samples simultaneously as the previous block is being emptied to the DAC?
4. Any external resource suggestions or explanation about how to work using real time block based DSP?

Thank you so much! I hope this post isn't asking too many questions at once