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Carey Carlan
 
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Default Combining different sample rates

Can you think of a way to combine these two sources without resampling?

I'm going to be copying some music from video tape recorded with a PCM-F1
to CD. Most of these tapes have ambient channels recorded on the HiFi
audio tracks of the video tape.

I've aquired a PCM-601ES unit with digital output, but I understand that
it's 44.05593846 kHz.

I'll sample the analog tracks through my Cranesong Spider (best converters
I own) at true 44.1 kHz.

The front source is 44.056K. I don't have another clock I can set to that
rate and the 601 has no word clock output, so there's no way to send both
streams synched and clocked. Am I right?

My computer will accept both streams simultaneously through separate SPDIF
and Lightpipe inputs. I can then resample the back channels to 44.056 and
then just tell the file it's 44.1 without resampling. I don't need sample
accurate alignment of the front and back channels, and I don't think the
slight pitch change (0.1%) will bother anyone.
  #2   Report Post  
Arny Krueger
 
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Carey Carlan wrote:

Can you think of a way to combine these two sources without
resampling?


Not if they have materially different sample rates.

If I was going to combine *anything* that was already in the digital
domain, I'd be very prone to do it with Audition/CE.

I'm going to be copying some music from video tape recorded with a
PCM-F1 to CD. Most of these tapes have ambient channels recorded on
the HiFi audio tracks of the video tape.


I've aquired a PCM-601ES unit with digital output, but I understand
that it's 44.05593846 kHz.


It misses 44.1 KHz by about 1%. It obviously needs to be resampled.

As close as I can get with Audition CE is 44056 Hz resampled to 44.1.
That is 0.00014 % off. Seems like close enough.

I'll sample the analog tracks through my Cranesong Spider (best
converters I own) at true 44.1 kHz.


OK

The front source is 44.056K. I don't have another clock I can set

to
that rate and the 601 has no word clock output, so there's no way to
send both streams synched and clocked. Am I right?


Seems like, but why worry?

My computer will accept both streams simultaneously through separate
SPDIF and Lightpipe inputs.


Why bother - why not read them one at a time and combine in the
digital domain?



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Scott Dorsey
 
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In article ,
Carey Carlan wrote:
Can you think of a way to combine these two sources without resampling?

I'm going to be copying some music from video tape recorded with a PCM-F1
to CD. Most of these tapes have ambient channels recorded on the HiFi
audio tracks of the video tape.

I've aquired a PCM-601ES unit with digital output, but I understand that
it's 44.05593846 kHz.

I'll sample the analog tracks through my Cranesong Spider (best converters
I own) at true 44.1 kHz.


This is bad. The thing is that SRC is very hard to do with sample rates
that are extremely similar. This is probably the worst case scenario
for SRC.

The front source is 44.056K. I don't have another clock I can set to that
rate and the 601 has no word clock output, so there's no way to send both
streams synched and clocked. Am I right?


Well, that's a question... if you COULD split the output of the 601 with
a DA, you would be able to use the S-PDIF signal as a reference clock.
The problem is just that it's going to be a lousy reference clock because
it'll be referenced to the data clock in the 601 which is not exactly good.

If you had an Apogee AD-1000, or some other converter that could sample
at 44.056, you could just use it to record the analogue tracks wild and
just expect that there might be a little drift here and there.

If you had a master reference clock that would lock to an external source,
like the Big Ben, I think you could use that to clean up the clock coming
out of the 601's S-PDIF connection before putting it into the Cranesong.

I think Audio Alchemy made a little reclocking box back in the late
eighties, and it sold for a couple hundred bucks when it was new. It took
a dirty S-PDIF input and give you a reasonably cleanly-clocked output.

The real question is how the Cranesong will deal with a lousy clock input,
and you should ask them that. They may have some solution that I haven't
thought of.

My computer will accept both streams simultaneously through separate SPDIF
and Lightpipe inputs. I can then resample the back channels to 44.056 and
then just tell the file it's 44.1 without resampling. I don't need sample
accurate alignment of the front and back channels, and I don't think the
slight pitch change (0.1%) will bother anyone.


Alternatively, you could just take the rear channels, which I assume
are on the analogue tracks and which probably don't sound all that
wonderful anyway, and just run them through the SRC and see what
happens. They can't get all THAT much worse.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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daz[at]roughdiamondmarketing[dot]com
 
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Default

why not record each at their own clock rate, save as 44.1 and then pitch
shift the wrong one ?

"Carey Carlan" wrote in message
. 191...
Can you think of a way to combine these two sources without resampling?

I'm going to be copying some music from video tape recorded with a PCM-F1
to CD. Most of these tapes have ambient channels recorded on the HiFi
audio tracks of the video tape.

I've aquired a PCM-601ES unit with digital output, but I understand that
it's 44.05593846 kHz.

I'll sample the analog tracks through my Cranesong Spider (best converters
I own) at true 44.1 kHz.

The front source is 44.056K. I don't have another clock I can set to that
rate and the 601 has no word clock output, so there's no way to send both
streams synched and clocked. Am I right?

My computer will accept both streams simultaneously through separate SPDIF
and Lightpipe inputs. I can then resample the back channels to 44.056 and
then just tell the file it's 44.1 without resampling. I don't need sample
accurate alignment of the front and back channels, and I don't think the
slight pitch change (0.1%) will bother anyone.



  #5   Report Post  
Bob Cain
 
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Default



Scott Dorsey wrote:

This is bad. The thing is that SRC is very hard to do with sample rates
that are extremely similar. This is probably the worst case scenario
for SRC.


True if done with polyphase filters but not if done by sinc
interpolation. I believe that Adobe Audition uses the
latter at its maximum quality setting.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein


  #6   Report Post  
Arny Krueger
 
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Default

daz[at]roughdiamondmarketing[dot]com wrote:

why not record each at their own clock rate, save as 44.1 and then
pitch shift the wrong one ?


In essence, that's what I recommended.


  #8   Report Post  
Scott Dorsey
 
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Carey Carlan wrote:
(Scott Dorsey) wrote in

Well, that's a question... if you COULD split the output of the 601
with a DA, you would be able to use the S-PDIF signal as a reference
clock. The problem is just that it's going to be a lousy reference
clock because it'll be referenced to the data clock in the 601 which
is not exactly good.


That's the best idea I've heard so far. The 601 clock is no worse than the
signal recorded on the video hifi audio tracks. A bit of jitter really
won't mess that sound up much.

What kind of DA do I need? Would an active video splitter do it? (got one
of those).


Try and it see. It will probably work. It will add clock jitter, though.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
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