Taperssection.com
Gear / Technical Help => Battery Boxes, Preamps, Mixers, ADCs, and Processors => Topic started by: Kush on December 26, 2009, 03:58:17 PM
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It's been a while since I used my Oade T+ mod UA-5 for an on-the-fly matrix. In fact I've only ever used it for 16 bit 44.1 kHz recordings with a JB3.
Tonight I have the opportunity to have onstage mics and have a sbd feed. What settings do I use on the UA-5 to make sure it's sending a 24 bit 48 kHz feed to my HD-P2?
Onstage mics > XLRs > UA-5 with +48v phantom power enabled.
sbd feed (tape out) > RCAs > UA-5 (adjust levels via RCA input volume knob on rear)
UA-5
recording source set to: analog
sample rate set to: 48 kHz
input monitor mode set to: stereo
output: cranked all the way
Hi-Z or Lo-Z?
rear switch set to: coaxial
ADV On or Off?
Coaxial S/PDIF out of UA-5 into digital in on the HD-P2
what should my settings on the HD-P2 be?
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It's been a while since I used my Oade T+ mod UA-5 for an on-the-fly matrix. In fact I've only ever used it for 16 bit 44.1 kHz recordings with a JB3.
Tonight I have the opportunity to have onstage mics and have a sbd feed. What settings do I use on the UA-5 to make sure it's sending a 24 bit 48 kHz feed to my HD-P2?
Onstage mics > XLRs > UA-5 with +48v phantom power enabled.
sbd feed (tape out) > RCAs > UA-5 (adjust levels via RCA input volume knob on rear)
UA-5
recording source set to: analog
sample rate set to: 48 kHz
input monitor mode set to: stereo
output: cranked all the way
Hi-Z or Lo-Z?
rear switch set to: coaxial
ADV On or Off?
Coaxial S/PDIF out of UA-5 into digital in on the HD-P2
what should my settings on the HD-P2 be?
If it's digi-modded like most/all of the others in the field, then it's set to 24bit regardless of what you stick behind it (and your JB3 is just chopping bits 17-24). If I remember correctly, the output is just your headphone output, so unless your monitoring or feeding that signal elsewhere, it doesn't matter. Also, based on the digi-mod side-effects (at least on the oade modded ones, can someone clarify if it applies to the general digi-mod instructions here?) then both outputs (coax and optical) are hot regardless of what you set the switch to.
I thought the Advanced switch was for 96khz rec/playback only?
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I just set everything up at home with my mics in front of my speakers and the RCAs coming out of my receiver and everything seemed to be working just fine and recording a 24 bit 48 kHz signal. Should have done that first and posted later.
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I just set everything up at home with my mics in front of my speakers and the RCAs coming out of my receiver and everything seemed to be working just fine and recording a 24 bit 48 kHz signal. Should have done that first and posted later.
no worries. People ran the UA-5s in what they thought was a true 16bit mode for years before someone with a 24bit recorder figured it out.
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With the gear that you mentioned, you could run:
mics>UA-5>JB3 (optical)
+
SBD>HDP2 (analog)
(or vice-versa SBD>UA-5>JB3 + mics>HDP2)
And then use the digi coax output of the UA-5 to feed the coax input of the HDP2 (you have to set the clock source of HDP2 to coax). So the clock source would be coming from the UA-5. The UA-5 only puts out 24bit, so no worries there. Both opti and coax outputs on UA-5 are hot at the same time.
Or, if you'd rather run an on-the-fly mix, then run:
mics>UA-5(xlr)
+
SBD>UA-5(rca)
>HDP2(coax)
Personally, I'd go the first option. Much more flexibility.
Run each rig individually, then sync em up in post. Since they are on the same clock source, syncing up is easy.
Hi-Z should be off.
ADV is for 24/96 recording only.
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I just set everything up at home with my mics in front of my speakers and the RCAs coming out of my receiver and everything seemed to be working just fine and recording a 24 bit 48 kHz signal. Should have done that first and posted later.
no worries. People ran the UA-5s in what they thought was a true 16bit mode for years before someone with a 24bit recorder figured it out.
Whoa! Am I reading this right...?
All these years - and jillions of UA5 > JB3 recordings are TRUNCATED??
How did that happen?
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I just set everything up at home with my mics in front of my speakers and the RCAs coming out of my receiver and everything seemed to be working just fine and recording a 24 bit 48 kHz signal. Should have done that first and posted later.
no worries. People ran the UA-5s in what they thought was a true 16bit mode for years before someone with a 24bit recorder figured it out.
Whoa! Am I reading this right...?
All these years - and jillions of UA5 > JB3 recordings are TRUNCATED??
How did that happen?
If I remember correctly, the word/bit length is set via the software interface/connection that normally exists and removing that link causes the device to stay in 24bit mode. I'm trying to find the thread, but TS.com's search function once again gets the better of me. Brian and some others spoke about it like 4 years ago.
edit: paging Brian, cleanup on aisle 4.
edit2: found it (http://taperssection.com/index.php?topic=90299.msg1202236;topicseen#msg1202236), original discussion/discovery A (http://taperssection.com/index.php?topic=86745.msg1168345#msg1168345) and B (http://taperssection.com/index.php?topic=86745.msg1171403#msg1171403).
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It probably doesn't matter if there was truncation--you'd have to have a combined ambient/electrical noise level lower than about -87dBFS for truncation to cause quantization distortion. For example, if you were peaking at say -10dBFS, and the peak-to-RMS was say 14dB, and the show was 105dBSPL, that means the acoustic noise floor would have to be less than 42dBSPL-ish. That isn't terribly likely doing concert recording. And even if the concert did have a lower noise floor, it's not like there is a switch that suddenly creates gobs of distortion, it's a continuum. It's really only very bad for large reductions in dynamic range.
yeah, it took people almost 5 years to figure it out, can't be that bad... ;D
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yeah, it took people almost 5 years to figure it out, can't be that bad... ;D
I thought about this a while ago... at least in my case, most of my UA-5>JB3/H120 tapes came into Audacity or equivalent, got converted to 32bit floating (default for my editor)... then I level/fade/trim, etc. When I saved it back to disk as 16 bit, it gets dithered per editor's algorithm. So most of my tapes got dithered at this step. I'm not sure if that's the norm for most people, if so, most of them aren't really truncated when they are seeded. I don't think truncating or dithering before the editor stage matters much, and in fact repetitive dithering isn't really ideal.
But I agree... I don't think that I can hear the difference, not with my ears, and my playback system.
And back to the original topic... I agree with Uncle. Run 2 separate feeds to different decks and sync them if you can. The ability to adjust the mix in post, and/or take out any delay, far exceeds whatever you might gain by recording 24bit.
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Mixing two synced sets of files in post will yield better results. When I ran the UA-5 > P2 set up the biggest down side was that neither headphone output had enough gain to really hear what you are doing with the mix. Even at a moderately loud show. Sometimes it made for a less than stellar mix.
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I don't think that I can hear the difference, not with my ears, and my playback system.
I agree, it's on my list of "cool things to aspire to" in general workflow, but I'll take truncation over analog tapes any day of the week.
And back to the original topic... I agree with Uncle. Run 2 separate feeds to different decks and sync them if you can. The ability to adjust the mix in post, and/or take out any delay, far exceeds whatever you might gain by recording 24bit.
ditto.