[ajs]

Regarding the problem of the snippet of audio that occurs at the end of a region when recorded with ADC is on... Here is some more info:

First of all, the problem only happens (for me at least) when no range is selected (the start and end point are the same) and I manually stop the recording. If I select a range, then the snippet does NOT occur even if I manually stop in the middle of the range during recording... ie, it works properly (Strange that someone posted the exact opposite of this a few months ago... don't know why).

Second, as I mentioned previously, the snippet that is stuck at the end is the audio that is located about 3 seconds prior to the end of the region (ie, 131072 samples prior to the snippet start). HOWEVER, if the recording is less than 3 seconds in length, the snippet comes from audio recorded in a PREVIOUS take. If it is the first recording after opening a session, then the snippet is blank.

Try this:
1. start PT, create new 44.1/24 session with 2 audio tracks, put a plug-in(s) with a reasonably large delay on the 2nd track, turn ADC on, rec-enable 1st track, select a start point (but not range), connect an instrument, let a note ring, hit record then hit stop within 2 seconds, all the while letting that note ring. Zoom in on the end of the region. You should see a range equal in size to the dly+cmp where there is silence, even though the note was ringing.

2. Now hold a note and record >3 seconds of audio, holding the note the entire time (again, do NOT select a range just a start point), then undo, silence your instrument, then record 2 seconds (or any length less than 3 seconds) of silence. Zoom in on the end of the supposedly silent region and you will see a snippet from the >3 second recording.

HERE'S MY THEORY:

PT is using an approx 3 second circular buffer to store the audio before writing to disk. Being "circular", when PT gets to the end of the buffer it starts writing to the beginning again. It reuses the buffer without clearing it between takes, leaving old audio in the buffer.

When ADC is on, PT is assuming the musician, who is playing to the delayed signal, is playing notes that are late by the max amount of delay (call it d samples), so when it reads from the buffer to write to disk, PT reads from sample location that is d samples later than the time it is writing to disk.

BUT... when a user who has sets a single start/end point hits STOP during a recording take, PT stops recording to the buffer immediately rather than recording another d samples. Since these d samples are written to disk at the end of a region, it is writing samples either from the previous recording take, or if the current take is longer than the 3 second buffer length, from 3 seconds prior to the end of the current recording!!! So if PT continued recording to the buffer for d samples after pressing STOP (as it should since the musician should be hearing everything d samples late), it wouldn't have the snippet from the previous recording left in the buffer.

ok, this is just my theory and I don't know anything about how the PT code works, but desperate times call for desperate measures!