lotus

previous page: 5.10 - What is PCM-F1 format?
  
page up: Audio Professional FAQ
  
next page: 5.12 - How can a 44.1 kHz sampling rate be enough to record all the harmonics of music? Doesn't that mean that we chop off all the harmonicsabove 20 khz? Doesn't this affect the music? After all, analog systemsdon't filter out all the information above 20 kHz, do they?

5.11 - How do digital recorders handle selective synchronization?




Description

This article is from the Audio Professional FAQ, by with numerous contributions by Gabe M. Wiener others.

5.11 - How do digital recorders handle selective synchronization?

Selective Synchronization, or "sel-sync" as it is often called, is the
ability of a recorder to play and record simultaneously, allowing
synchronous recording of new material onto specific tracks without
erasing everything on tape. This technique is what makes overdubbing
possible.

On an analog recorder, audio tracks are discrete entities, and the
sync head is really just a stack of individual heads, any one of which
is capable of recording or playing back. Thus sel-sync is a
relatively simple matter of putting some heads into record and others
into repro.

In the digital world, the problem is highly complex. First, A/D and
D/A conversion involves an acquisition delay of several milliseconds.
Second, and more importantly, digital tracks are not discrete. Rather,
they are multiplexed together on a tape, along with subcode and other
non-audio information. So how can you replace one track and leave the
others untouched?

The answer is a technique called "read before write" (RBW) or "read,
modify, write" (RMW) which involves a second set of heads. The data
is read from the tape and flushed into a buffer, where it can be
modified, and ultimately written back to the tape. Thus when you
"punch in" on a digital deck, you are physically re-writing all the
tracks, not just the one you're overdubbing. You are not, however,
changing the data on any track other than the one you want to
replace. [Gabe]


 

Continue to:













TOP
previous page: 5.10 - What is PCM-F1 format?
  
page up: Audio Professional FAQ
  
next page: 5.12 - How can a 44.1 kHz sampling rate be enough to record all the harmonics of music? Doesn't that mean that we chop off all the harmonicsabove 20 khz? Doesn't this affect the music? After all, analog systemsdon't filter out all the information above 20 kHz, do they?