A question came up, and as a layman I am not sure how to appropriately find an answer; bear with me.
Given a 2 minute audio file, say a .wav file sampled at 44,100 Hz, I am curious if it is possible to effectively convert this audio file into a 4 minute one "sampled" at 22,050 Hz via the following process.
We have 44,100 blocks of data being stored every second. It stands to reason that a 22,050 Hz sampler expects half as many blocks per second. One approach to match this slower request rate would be to simply feed in every other block from the original sampling, but this seems like a highly lossy process and wouldn't extend the length of the audio file (also curious to know what effect this would have on this pitch, if any). On the other hand, we can match the slower request rate by feeding in each of the blocks one by one--just half as fast.
In my estimation, feeding the blocks in half as fast means in would take 4 minutes instead of two to finish feeding in the blocks.
Is this a valid strategy to produce a new .wav file?
Would this new .wav file be playable in any meaningful way?
Would the pitch of this wave file be lower?
What is the limiting behavior of iterating this process (Or, what does a .wav file sampled at 1 Hz sound like)?