Topic: microtrack records at 44.1 from a 48 source??? please help with this one.

[KLowe]

Alright.  I'll do my best to explain this one.  I was taping Umphs at the sonic stage at Bonnaroo.  Had UA-5 set to 24/48...taped show and all good.
here is the F'ed up thing.

When I go to listen to the recording fromt the MT it is in 24/44.1 .  Really weird thing is the "speed" of the show is slowed down and the voices distorted.  (like playing a record to slow).  Now I KNOW that the UA-5 was set on 48 b/c the JB3 back-up that I was running picked up and recorded the 48 signal with no worries.

When I go to set the sample rate on the MT is says (auto) and cannot be changed to anything.  Obviously "auto" sucks and it recorded to wrong sample rate. 

here is the question.  Did I do something wrong?  Is there a special setting I need to use when recording SPDIF "in"? 

Running the latest firmware.

Thanks in advance for any help. 

[udovdh]

What firmware are you using?
Was the s/pdif signal active before you started the MT?

[gngrbrdman13]

Alright.  I'll do my best to explain this one.  I was taping Umphs at the sonic stage at Bonnaroo.  Had UA-5 set to 24/48...taped show and all good.
here is the F'ed up thing.

When I go to listen to the recording fromt the MT it is in 24/44.1 .  Really weird thing is the "speed" of the show is slowed down and the voices distorted.  (like playing a record to slow).  Now I KNOW that the UA-5 was set on 48 b/c the JB3 back-up that I was running picked up and recorded the 48 signal with no worries.

When I go to set the sample rate on the MT is says (auto) and cannot be changed to anything.  Obviously "auto" sucks and it recorded to wrong sample rate. 

here is the question.  Did I do something wrong?  Is there a special setting I need to use when recording SPDIF "in"? 

Running the latest firmware.

Thanks in advance for any help. 

I have used borrowed MT units on the 'auto' setting with no worries.  You obviously had it set to spdif or would have nothing.  

Have you run a test at home to recreate the 'issue'?  Give that a try and report back.  


dd

[gngrbrdman13]

Was the s/pdif signal active before you started the MT?

Excellent question.  Skalinder taught me to have the signal running and then activate power on to the mt.  I actually ran the unit both ways to 'test' it and found that if the signal wasn't running and the power was activated on the mt then I activated my a/d that it didn't work.

When i powered down the mt and powered up the a/d then the mt it worked great.

Thanks for the advice Brian!   -T

dd

[KLowe]

ah ha.... I'm not sure about the signal thing...but it makes sense.  I bet thats what I did wrong.  I'll play around with it and see what happens.

Thanks for the help yall.
T's all around.

[Brian Skalinder]

Really weird thing is the "speed" of the show is slowed down and the voices distorted.  (like playing a record to slow).

I don't know if you've done this already, but FWIW you can resolve this problem in post very easily.  Just use your favorite WAV editor to open the file and change the WAV header's sample rate.  Note:  this is -not- resampling and should be a very quick operation.  If the editor needs to crunch for a while, then you're probably actually resampling which you do not want to do.

For example, in Adobe Audition:

• Adjust Sample Rate modifies the WAV header's sample rate entry without resampling
• Convert Sample Type actually performs sample rate conversion, i.e. it resamples

[KLowe]

Thanks Brian,
I'll give that a try.  (although trying to conceptualize what it is doing is making me just a bit dumber at the moment)


I'll let ya know.

[KLowe]

Skalinder...You are a damn Genious!!!   

now...since I have adjusted it to 48 (and it sounds normal now)...Do i need to resample convert to be able to transfer the file to othe devices so that it will play normally?

Now I don't hate my micrtrack....as much.

[udovdh]

now...since I have adjusted it to 48 (and it sounds normal now)...Do i need to resample convert to be able to transfer the file to othe devices so that it will play normally?

Do you understand what you did?
Do you understand what other devices (like what?) need? A CD-player needs 44.1 Khz...

Now I don't hate my micrtrack....as much.

It all depends on perspective and validity of reasons.

[KLowe]

Its good now.  I think I understand what I did...just changed the file header so the MT will playback at the correct sample rate.

Your MT statement is correct.  I guess the reasons I hate it have all been user errors....but it is still easier to blame it on the MT.

Thanks all.

[Brian Skalinder]

Skalinder...You are a damn Genious!!!   

Nah, I simply experienced the same problem previously, that's all.  And someone else here taught me about the WAV header sample rate not matching the data sample rate, so...whoever that was (no chance of remembering), thank them. 

now...since I have adjusted it to 48 (and it sounds normal now)...Do i need to resample convert to be able to transfer the file to othe devices so that it will play normally?

Just depends on what other devices you want to play the recording.

Its good now.  I think I understand what I did

There are two sample rate elements involved here.  There's the sample rate of the actual data, i.e. how many samples of data per second are actually recorded.  And then there's the sample rate element in the WAV header that the playback device reads so it knows what sample rate to use during playback.  These sample rate elements need to match one another.  If they do not, then upon playback the recording will sound either too fast or too slow, depending on how the elements are mismatched.  Occasionally, some devices incorrectly write a WAV header sample rate element that does not match the actual data.  All you've done here is change the WAV header sample rate element so it matches the actual data.

I guess the reasons I hate it have all been user errors....but it is still easier to blame it on the MT.

I wouldn't say it's user error - the MT should lock onto the incoming signal properly and write the WAV header properly.  But, the fact is, that doesn't always happen.  And it's not the first, or only, device to suffer from this problem.  As someone else mentioned already, I've found feeding the MT a live S/PDIF signal before powering up helps prevent at least several quirks I encountered fairly regularly.

[Sid Viscous]

Really weird thing is the "speed" of the show is slowed down and the voices distorted.  (like playing a record to slow).

I don't know if you've done this already, but FWIW you can resolve this problem in post very easily.  Just use your favorite WAV editor to open the file and change the WAV header's sample rate.  Note:  this is -not- resampling and should be a very quick operation.  If the editor needs to crunch for a while, then you're probably actually resampling which you do not want to do.

For example, in Adobe Audition:

• Adjust Sample Rate modifies the WAV header's sample rate entry without resampling
• Convert Sample Type actually performs sample rate conversion, i.e. it resamples

The file needs to be shortened and the pitch changed. Wavelab can do this easily with fairly high quality, but it will never be as good as the right sample rate would have been.

[gngrbrdman13]

Skalinder...You are a damn Genious!!!   

Jesus his ego is big enough without you saying shit like that

[Brian Skalinder]

The file needs to be shortened and the pitch changed. Wavelab can do this easily with fairly high quality, but it will never be as good as the right sample rate would have been.

I don't follow.  Changing the WAV header so its sample rate matches the actual sample rate of the data does not require <a> shortening the file or <b> changing pitch, and it leaves the data itself intact with no modifications.  IME it sounds precisely as good as if the recorder had written the proper sample rate to the WAV header in the first place - why wouldn't it, we haven't changed the data at all, just the WAV header sample rate setting.

[Sid Viscous]

The file needs to be shortened and the pitch changed. Wavelab can do this easily with fairly high quality, but it will never be as good as the right sample rate would have been.

I don't follow.  Changing the WAV header so its sample rate matches the actual sample rate of the data does not require <a> shortening the file or <b> changing pitch, and it leaves the data itself intact with no modifications.  IME it sounds precisely as good as if the recorder had written the proper sample rate to the WAV header in the first place - why wouldn't it, we haven't changed the data at all, just the WAV header sample rate setting.

I gave a way to deal with the problem. You could always explain your method instead of being a dick.