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A new way to screw up a take.

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Anonymous
November 22, 2004 2:55:19 AM

Archived from groups: rec.audio.pro (More info?)

In my rack, the monitor mixer slides out on a shelf just below the 24 track
hard disk recorder (SDR 24/96).

I pulled the headphone cable, inserted a headphone extension cord and
bumped the rewind button, then stretched out across the room, listening
without realizing that I wasn't recording anything. Never done that
before.

These new computer recorders have such light touch buttons that just
brushing against them activates the function.

The worst part is it took me until just now (hours later) to realize how I
lost that take.

More about : screw

Anonymous
November 22, 2004 2:55:20 AM

Archived from groups: rec.audio.pro (More info?)

On Sun, 21 Nov 2004 18:55:19 -0500, Carey Carlan wrote
(in article <Xns95A8C0840C811gulfjoehotmailcom@207.69.189.191>):

> In my rack, the monitor mixer slides out on a shelf just below the 24 track
> hard disk recorder (SDR 24/96).
>
> I pulled the headphone cable, inserted a headphone extension cord and
> bumped the rewind button, then stretched out across the room, listening
> without realizing that I wasn't recording anything. Never done that
> before.
>
> These new computer recorders have such light touch buttons that just
> brushing against them activates the function.
>
> The worst part is it took me until just now (hours later) to realize how I
> lost that take.

I feel your pain. My favorite is forgetting to reset the sample rate on the
external A/D converter. I can actually do a 44.1 session in Pro Tools with
the External clock set to 48 and never hear anything weird until I playback
the mix to disk master.

I have rescued the mix by sending it out to a DAT machine analogoly and
recording it to 44.1, then reimporting it. But the anything else is screwed.

Regards,

Ty Ford



-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 22, 2004 2:55:21 AM

Archived from groups: rec.audio.pro (More info?)

Ty Ford wrote:
>
> forgetting to reset the sample rate on the
> external A/D converter. I can actually do a 44.1 session in Pro Tools with
> the External clock set to 48 and never hear anything weird until I playback
> the mix to disk master.
>
> I have rescued the mix by sending it out to a DAT machine analogoly and
> recording it to 44.1, then reimporting it. But the anything else is screwed.

Surely the PT hardware locks to the incoming signal and doesn't resample. In that case, you should be able to correct the headers on the WAV files and all should be well.

<http://railjonrogut.com/HeaderInvestigator.htm&gt; will do the job nicely.
Related resources
Anonymous
November 22, 2004 2:55:22 AM

Archived from groups: rec.audio.pro (More info?)

On Sun, 21 Nov 2004 22:21:54 -0500, Kurt Albershardt wrote
(in article <30d4agF2voeccU2@uni-berlin.de>):

> Ty Ford wrote:
>>
>> forgetting to reset the sample rate on the
>> external A/D converter. I can actually do a 44.1 session in Pro Tools with
>> the External clock set to 48 and never hear anything weird until I playback
>> the mix to disk master.
>>
>> I have rescued the mix by sending it out to a DAT machine analogoly and
>> recording it to 44.1, then reimporting it. But the anything else is screwed.
>
> Surely the PT hardware locks to the incoming signal and doesn't resample. In

> that case, you should be able to correct the headers on the WAV files and all

> should be well.
>
> <http://railjonrogut.com/HeaderInvestigator.htm&gt; will do the job nicely.

hmm seems to be for PC only. Drat!

Ty





-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 22, 2004 5:26:40 AM

Archived from groups: rec.audio.pro (More info?)

Carey Carlan <gulfjoe@hotmail.com> wrote in
news:Xns95A8C0840C811gulfjoehotmailcom@207.69.189.191:

> In my rack, the monitor mixer slides out on a shelf just below the 24
> track hard disk recorder (SDR 24/96).
>
> I pulled the headphone cable, inserted a headphone extension cord and
> bumped the rewind button, then stretched out across the room, listening
> without realizing that I wasn't recording anything. Never done that
> before.
>
> These new computer recorders have such light touch buttons that just
> brushing against them activates the function.
>
> The worst part is it took me until just now (hours later) to realize how
> I lost that take.

We had one guy that was so bad we installed a lockout. Once in record,
one had to hit stop and rewind at the same time to stop recording.
Hitting any single button while in record did nothing save for pause.
Pause was far enough away from the rest of the buttons that it wasn't a
problem.

r

--
Nothing beats the bandwidth of a station wagon filled with DLT tapes.
Anonymous
November 22, 2004 8:09:54 PM

Archived from groups: rec.audio.pro (More info?)

<< hmm seems to be for PC only. Drat!

Ty >>



You can do it with the Mac too. You need a file editing proram to do it like
older versions of Norton utilities that contain "Disk Editor", or something
called ResEdit, made by Apple I think. There's a line in SDII files that
indicates the sample rate that can be rewritten to the desired rate, solving
the problem.


Ted Spencer, NYC

"No amount of classical training will ever teach you what's so cool about
"Tighten Up" by Archie Bell And The Drells" -author unknown
Anonymous
November 22, 2004 8:09:55 PM

Archived from groups: rec.audio.pro (More info?)

"Ted Spencer" <prestokid@aol.com> wrote in message
news:20041122120954.23417.00001022@mb-m03.aol.com...
> << hmm seems to be for PC only. Drat!
>
> Ty >>


>
> You can do it with the Mac too. You need a file editing proram to do it
> like
> older versions of Norton utilities that contain "Disk Editor", or
> something
> called ResEdit, made by Apple I think. There's a line in SDII files that
> indicates the sample rate that can be rewritten to the desired rate,
> solving
> the problem.

Adobe Audition (CoolEditPro) has this feature built in. It has proven
useful a few times, when I did exactly what Ty described.

Steve King
Anonymous
November 22, 2004 10:19:33 PM

Archived from groups: rec.audio.pro (More info?)

On Mon, 22 Nov 2004 12:09:54 -0500, Ted Spencer wrote
(in article <20041122120954.23417.00001022@mb-m03.aol.com>):

> << hmm seems to be for PC only. Drat!
>
> Ty >>


>
> You can do it with the Mac too. You need a file editing proram to do it like
> older versions of Norton utilities that contain "Disk Editor", or something
> called ResEdit, made by Apple I think. There's a line in SDII files that
> indicates the sample rate that can be rewritten to the desired rate, solving
> the problem.
>
>
> Ted Spencer, NYC
>
> "No amount of classical training will ever teach you what's so cool about
> "Tighten Up" by Archie Bell And The Drells" -author unknown

And that wouldn't change the pitch or run time? Thanks code warrior!

Ty



-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 22, 2004 10:19:34 PM

Archived from groups: rec.audio.pro (More info?)

"Ty Ford" <tyreeford@comcast.net> wrote in message
news:LeOdnewXTtiIHz_cRVn-hw@comcast.com...
> On Mon, 22 Nov 2004 12:09:54 -0500, Ted Spencer wrote
> (in article <20041122120954.23417.00001022@mb-m03.aol.com>):
>
>> << hmm seems to be for PC only. Drat!
>>
>> Ty >>


>>
>> You can do it with the Mac too. You need a file editing proram to do it
>> like
>> older versions of Norton utilities that contain "Disk Editor", or
>> something
>> called ResEdit, made by Apple I think. There's a line in SDII files that
>> indicates the sample rate that can be rewritten to the desired rate,
>> solving
>> the problem.
>>
>>
>> Ted Spencer, NYC
>>
>> "No amount of classical training will ever teach you what's so cool about
>> "Tighten Up" by Archie Bell And The Drells" -author unknown
>
> And that wouldn't change the pitch or run time? Thanks code warrior!
>
> Ty
>
If I understand correctly, the problem is that the file has a header that is
inconsistent with its actual parameters. Therefore, it will play
improperly... either faster or slower than it should. By editing the
header, the computer will recognize it for what it is and it will play
properly. Correct pitch and correct length. It may be 48 kHz and you want
44.1, or it may be 24 bit and you want 16... but fixing that requires a
sample rate conversion process.

Steve King
Anonymous
November 23, 2004 7:00:01 AM

Archived from groups: rec.audio.pro (More info?)

>And that wouldn't change the pitch or run time?

That's what you want to have happen. The pitch/run time is already wrong due to
the playback sample rate being different than the file's data rate.

For example if you recorded a 44.1 file into a system that was clocking at 48K,
it will play back normally as long as it remains clocked at 48. When you play
it back at 44.1, it will be slow. That's because it contains a 48K data header
within the file, and it's now being clocked at 44.1. Changing the header back
to 44.1 speeds it back up again, undoing the problem.


Ted Spencer, NYC

"No amount of classical training will ever teach you what's so cool about
"Tighten Up" by Archie Bell And The Drells" -author unknown
Anonymous
November 23, 2004 8:59:10 PM

Archived from groups: rec.audio.pro (More info?)

prestokid@aol.com (Ted Spencer) wrote in message news:<20041122120954.23417.00001022@mb-m03.aol.com>...
> << hmm seems to be for PC only. Drat!
>
> Ty >>


>
> You can do it with the Mac too. You need a file editing proram to do it like
> older versions of Norton utilities that contain "Disk Editor", or something
> called ResEdit, made by Apple I think. There's a line in SDII files that
> indicates the sample rate that can be rewritten to the desired rate, solving
> the problem.
>
>
> Ted Spencer, NYC
>
> "No amount of classical training will ever teach you what's so cool about
> "Tighten Up" by Archie Bell And The Drells" -author unknown

In Digital Performer 3.x you can change the session sample rate to the
correct one (e.g., 48kHz) and perform the "merge soundbites" operation
and presto, the files are rewritten at the correct sample rate. You'd
think it would try to do some screwy sample rate conversion thing, but
it doesn't, it just copies the data to new files with the correct
header. Not as elegant as changing the header directly, but it's
saved my ass a few times...

Andrew Leavitt
Anonymous
November 25, 2004 2:46:17 AM

Archived from groups: rec.audio.pro (More info?)

On Mon, 22 Nov 2004 23:00:01 -0500, Ted Spencer wrote
(in article <20041122230001.21555.00001093@mb-m11.aol.com>):

>> And that wouldn't change the pitch or run time?
>
> That's what you want to have happen. The pitch/run time is already wrong due
> to
> the playback sample rate being different than the file's data rate.
>
> For example if you recorded a 44.1 file into a system that was clocking at
> 48K,
> it will play back normally as long as it remains clocked at 48. When you play
> it back at 44.1, it will be slow. That's because it contains a 48K data
header
> within the file, and it's now being clocked at 44.1. Changing the header back
> to 44.1 speeds it back up again, undoing the problem.

Ted,

The weird thing is, it's fine during playback and even mix to disk, but when
I play back the results of the mixed to disk "master" the pitch is qwangst. I
think the time is too.

What I'd want to be able to do is correct the pitch and the time.

Regards.

Ty

-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 25, 2004 6:03:15 PM

Archived from groups: rec.audio.pro (More info?)

Ty Ford wrote:

>The weird thing is, it's fine during playback and even mix to disk, but when
>I play back the results of the mixed to disk "master" the pitch is qwangst. I
>
>think the time is too.
>
>What I'd want to be able to do is correct the pitch and the time.
>

"Qwangst"...LOL...

We are talking about the problem of a real time digital copy having been made
into a host system at the wrong sample rate, right? The usual scenario is:

A 44.1 DAT or CD is played from its own transport in real time into a digital
interface on a DAW with the software unwittingly set to internally clock at 48K
(not all systems will allow this; of mine, only the Audiomedia II card will).
The material goes in fine and plays back fine once it's there. Then you mix to
disk and burn it to a CD. Oops! It plays back at the wrong speed/length. If you
rate convert it you then have a 44.1 version that *still* plays back at the
wrong speed.

Right?

If that's the setup/problem, then changing the sample rate header is indeed the
fix. It will change the file back to 44.1 *and* force it to play back at the
correct speed/time.

Another poster mentioned a trick in DP that I was unaware of, and that a brief
test I did seemed to bear out. It involves importing the miscreant 48K file
"illegally" into a track in a 44.1 session and using "merge soundbites" on it
from the edit menu. It reburns the file at 44.1 and correspondingly corrects
its speed/time. It appears to be an undocumented (unintended?) feature, but
seemingly works. If so, it might be a more accessible solution than editing the
file header.


Ted Spencer, NYC

"No amount of classical training will ever teach you what's so cool about
"Tighten Up" by Archie Bell And The Drells" -author unknown
Anonymous
November 26, 2004 12:52:19 PM

Archived from groups: rec.audio.pro (More info?)

On Thu, 25 Nov 2004 10:03:15 -0500, Ted Spencer wrote
(in article <20041125100315.06154.00000708@mb-m21.aol.com>):

> Ty Ford wrote:
>
>> The weird thing is, it's fine during playback and even mix to disk, but
>> when
>> I play back the results of the mixed to disk "master" the pitch is qwangst.
>> I
>>
>> think the time is too.
>>
>> What I'd want to be able to do is correct the pitch and the time.
>>
>
> "Qwangst"...LOL...

Ted,

I use the term qwangst with some trepidation. It was used in an audio
discussion years ago and when I asked for a definition I was told it meant
F**d up. Perhaps you could add something to that.
>
> We are talking about the problem of a real time digital copy having been made
> into a host system at the wrong sample rate, right? The usual scenario is:
>
> A 44.1 DAT or CD is played from its own transport in real time into a digital
> interface on a DAW with the software unwittingly set to internally clock at
> 48K
> (not all systems will allow this; of mine, only the Audiomedia II card will).
> The material goes in fine and plays back fine once it's there. Then you mix
to
> disk and burn it to a CD. Oops! It plays back at the wrong speed/length. If
> you
> rate convert it you then have a 44.1 version that *still* plays back at the
> wrong speed.
>
> Right?

Pretty much. After making a multi-track production in Pro Tools, you
usually do a Mix To Disk to make a stereo master. Occasionally, I have
forgotten to switch my external A/D converters to the same sample rate as the
production I'm working on.

So the clocks are set to 48, but the production is set to 44.1.

Pro Tools will not complain. I don't find out about it until I listen to the
stereo master, which will be of pitch and off speed.
>
> If that's the setup/problem, then changing the sample rate header is indeed
> the
> fix. It will change the file back to 44.1 *and* force it to play back at the
> correct speed/time.

Hmm, now all I need to do is figure out how to find the right info using that
Mac app.
>
> Another poster mentioned a trick in DP that I was unaware of, and that a
brief
> test I did seemed to bear out. It involves importing the miscreant 48K file
> "illegally" into a track in a 44.1 session and using "merge soundbites" on it
> from the edit menu. It reburns the file at 44.1 and correspondingly corrects
> its speed/time. It appears to be an undocumented (unintended?) feature, but
> seemingly works. If so, it might be a more accessible solution than editing
> the
> file header.

right.

Ty




-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 26, 2004 12:57:00 PM

Archived from groups: rec.audio.pro (More info?)

Ty Ford wrote:
>
> After making a multi-track production in Pro Tools, you
> usually do a Mix To Disk to make a stereo master. Occasionally, I have
> forgotten to switch my external A/D converters to the same sample rate as the
> production I'm working on.
>
> So the clocks are set to 48, but the production is set to 44.1.
>
> Pro Tools will not complain. I don't find out about it until I listen to the
> stereo master, which will be of pitch and off speed.

In that case, you have a 48k file that is marked in the headers as 44k1. To use it in a 44k1 session, you'll need to eidt the headers and SRC it down to 44k1.
Anonymous
November 27, 2004 2:39:42 AM

Archived from groups: rec.audio.pro (More info?)

On Fri, 26 Nov 2004 12:57:00 -0500, Kurt Albershardt wrote
(in article <30p93aF33ngbuU2@uni-berlin.de>):

> Ty Ford wrote:
>>
>> After making a multi-track production in Pro Tools, you
>> usually do a Mix To Disk to make a stereo master. Occasionally, I have
>> forgotten to switch my external A/D converters to the same sample rate as
>> the
>> production I'm working on.
>>
>> So the clocks are set to 48, but the production is set to 44.1.
>>
>> Pro Tools will not complain. I don't find out about it until I listen to
>> the
>> stereo master, which will be of pitch and off speed.
>
> In that case, you have a 48k file that is marked in the headers as 44k1. To
> use it in a 44k1 session, you'll need to eidt the headers and SRC it down to
> 44k1.

Hmm, yep, I figured it was at least a two step process. Thanks Kurt.

Ty




-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
November 27, 2004 8:04:36 PM

Archived from groups: rec.audio.pro (More info?)

Another problem is: Say your doing a :30 tv/radio spot, and your
daw/sequencer is set at 48k, but your ext clock is at 44.1k. The
sequencer and it's displayed timer will be running slower than it
thinks it is, making your perfect out at :30 really at :33 (or
whatever). Now your looking at time compression too. "Man, I thought
it felt slow". Shouldn't have thrown the old stop watch away.

I've only had to fix this kinda thing a couple of times.
Back in OS9 I used ResEdit or Soundhack (which will do pitch change
too) to change file headers. These would only do one file at a time,
but there are some that'll do it in batches.



Andrew Leavitt wrote:
> prestokid@aol.com (Ted Spencer) wrote in message news:<20041122120954.23417.00001022@mb-m03.aol.com>...
>
>><< hmm seems to be for PC only. Drat!
>>
>>Ty >>


>>
>>You can do it with the Mac too. You need a file editing proram to do it like
>>older versions of Norton utilities that contain "Disk Editor", or something
>>called ResEdit, made by Apple I think. There's a line in SDII files that
>>indicates the sample rate that can be rewritten to the desired rate, solving
>>the problem.
>>
>>
>>Ted Spencer, NYC
>>
>>"No amount of classical training will ever teach you what's so cool about
>>"Tighten Up" by Archie Bell And The Drells" -author unknown
>
>
> In Digital Performer 3.x you can change the session sample rate to the
> correct one (e.g., 48kHz) and perform the "merge soundbites" operation
> and presto, the files are rewritten at the correct sample rate. You'd
> think it would try to do some screwy sample rate conversion thing, but
> it doesn't, it just copies the data to new files with the correct
> header. Not as elegant as changing the header directly, but it's
> saved my ass a few times...
>
> Andrew Leavitt
Anonymous
November 28, 2004 11:49:00 AM

Archived from groups: rec.audio.pro (More info?)

On Sat, 27 Nov 2004 12:04:36 -0500, Bryson wrote
(in article <Es2qd.7480$Ua.4248@newsread3.news.atl.earthlink.net>):

> Another problem is: Say your doing a :30 tv/radio spot, and your
> daw/sequencer is set at 48k, but your ext clock is at 44.1k. The
> sequencer and it's displayed timer will be running slower than it
> thinks it is, making your perfect out at :30 really at :33 (or
> whatever). Now your looking at time compression too. "Man, I thought
> it felt slow". Shouldn't have thrown the old stop watch away.
>
> I've only had to fix this kinda thing a couple of times.
> Back in OS9 I used ResEdit or Soundhack (which will do pitch change
> too) to change file headers. These would only do one file at a time,
> but there are some that'll do it in batches.

Yes that's the real rub. The sample rate conflicts corrupt the time base of
the production.

Ty

-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
November 29, 2004 4:53:10 PM

Archived from groups: rec.audio.pro (More info?)

Bryson <redbugg@mindNOSPAMspring.com> wrote :

> Another problem is: Say your doing a :30 tv/radio spot, and your
> daw/sequencer is set at 48k, but your ext clock is at 44.1k. The
> sequencer and it's displayed timer will be running slower than it
> thinks it is, making your perfect out at :30 really at :33 (or
> whatever). Now your looking at time compression too. "Man, I thought
> it felt slow". Shouldn't have thrown the old stop watch away.
>


Yup, the BPMs get all screwed up, too, if you're working with a click
that's generated within the DAW. That's a good sign that something's
amiss if the tempos seem strangly faster or slower than they did
during pre-production...

Actually the first time I ran into this problem the odd seeming tempos
("boy, 95 seems really slow today...") tipped me off. When I realized
what was up I told the producer about it and started doing math to
figure out what the tempos actually were. I was sweating it, thinking
I'd totally blown it, but later he (the producer) told me that he was
really impressed that I'd informed him straight out about my screw up
and that I'd already figured out how to deal with it. Apparently
writing a few equations down on a piece of paper and pulling out a
calculator really impresses people... This was before I'd figured out
that Digital Performer will fix this problem with Merge Soundbites, so
after the sessions we (another engineer and I) routed the files (in
groups of 8) via ADAT lightpipe to a laptop with a MOTU 828 which was
set to the real sample rate. A real kludge, but it worked. Now I'm
pretty careful about checking that everything's at the right sample
rate...
!