Taperssection.com
Gear / Technical Help => Recording Gear => Topic started by: Rairun on April 23, 2023, 11:20:49 AM
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https://deitymic.com/products/pr-2/
https://proav.co.uk/audio/deity-pr-2-pocket-audio-recorder
This is not out yet, but if it is as good as it sounds like (-132dBV self-noise @ +30dB, 32-bit float, PiP of 3V / 5V / LINE Switchable, 30h on 2 AA batteries, very small), this is going to be the ultimate recorder for stealth taping. The only thing I don't like is that it seems to be 48 kHz only (I prefer to record at 44 kHz), but I'd be more than willing to overlook this for the other features.
In my experience, my CA-11 cards are around 16 dB quieter than the internal microphones of the Zoom H1 and the Roland R-05. They've never ever overloaded either of those recorders with a battery box, and if I decide to use my CA preamp, I can safely use a +15dB setting on the preamp if I set either of those recorders to unity gain (Zoom H1 at level 16, Roland R-05 using Line-in at level 40). Between the pre-amp and the recorders, I find that I usually use between +25dB gain (for loud shows) and +15db gain (for extremely loud shows).
So up to +30dB gain with such low self-noise (and 5V PiP to boot! might even be able to skip the battery box), with 32-float recording, sounds absolutely perfect.
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I'm interested...it looks tiny. This + DPA lavs would be the ultimate micro setup, and no need to attach a phone like DPA's MMA-A.
Any word on pricing?
How is their preamp quality compared to the usual suspects (sound devices)?
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https://youtu.be/yMdLNjIyRT4
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No word on pricing. The fact that it's timecode enabled will probably bably raise that price a bit. Otherwise, it's pretty much what we've waited for.
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This looks like it has the potential to be THE definitive stealth recorder.
Heck, with an outboard preamp it seems like it might even be fine for a two-channel rig in open taping scenarios.
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https://deitymic.com/products/pr-2/
https://proav.co.uk/audio/deity-pr-2-pocket-audio-recorder
This is not out yet, but if it is as good as it sounds like (-132dBV self-noise @ +30dB, 32-bit float, PiP of 3V / 5V / LINE Switchable, 30h on 2 AA batteries, very small), this is going to be the ultimate recorder for stealth taping. The only thing I don't like is that it seems to be 48 kHz only (I prefer to record at 44 kHz), but I'd be more than willing to overlook this for the other features.
In my experience, my CA-11 cards are around 16 dB quieter than the internal microphones of the Zoom H1 and the Roland R-05. They've never ever overloaded either of those recorders with a battery box, and if I decide to use my CA preamp, I can safely use a +15dB setting on the preamp if I set either of those recorders to unity gain (Zoom H1 at level 16, Roland R-05 using Line-in at level 40). Between the pre-amp and the recorders, I find that I usually use between +25dB gain (for loud shows) and +15db gain (for extremely loud shows).
So up to +30dB gain with such low self-noise (and 5V PiP to boot! might even be able to skip the battery box), with 32-float recording, sounds absolutely perfect.
Need more research on this AI smart level feature...Not so sure I would want that for taping.
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I'm interested...it looks tiny. This + DPA lavs would be the ultimate micro setup, and no need to attach a phone like DPA's MMA-A.
Any word on pricing?
How is their preamp quality compared to the usual suspects (sound devices)?
Yes the 5v cuts the need for a battery box. I’m interested.
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They haven't set the price yet, so don't appear too interested in case they are reading here... :yack:
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It records in stereo? I feel like these small body pack recorders are sometimes mono
EDIT: watched the video. Yes stereo recorder. Looks nice!
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I can forsee a use case for this. If I had it today, I'd have a use for it tomorrow :)
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I'm interested...it looks tiny. This + DPA lavs would be the ultimate micro setup, and no need to attach a phone like DPA's MMA-A.
Any word on pricing?
How is their preamp quality compared to the usual suspects (sound devices)?
Yes the 5v cuts the need for a battery box. I’m interested.
Are we certain it will provide +5 to both sides of a unbalanced stereo line in?
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looks interesting
they said in one of the videos that it would be available in july, yet there's still no pricing
wonder if it will be vaporware like their last 2-ch recorder?
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the DPA D:VICE is so small it fits into my coin compartment of my wallet - nobody will ever check that. super easy so smuggle in.
also you do not need to hide your phone, it raises no suspicion when looking at it and the phones battery lasts for hours for recording 24 bit.
here you will have an extra device with alcaline batteries (that can be connected remotely to - your phone).
unless the recording quality isnt muuuuch better than the DPA D:VICE i don't think this is much of a progress for stealth taping?
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the DPA D:VICE is so small it fits into my coin compartment of my wallet - nobody will ever check that. super easy so smuggle in.
also you do not need to hide your phone, it raises no suspicion when looking at it and the phones battery lasts for hours for recording 24 bit.
here you will have an extra device with alcaline batteries (that can be connected remotely to - your phone).
unless the recording quality isnt muuuuch better than the DPA D:VICE i don't think this is much of a progress for stealth taping?
yes but the cheapest d:vice+4061s is like $1400
if this is a few hundred dollars as expected its a $500 rig with the numerous cheap plug in power mics
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With a babynbox and Schoeps I would be intrigued.
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Competition:
- https://lectrosonics.com/spdr-stereo-personal-digital-recorder.html
- Tentacle: not yet
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the lectrosonic one is certainly pricey...
https://www.bhphotovideo.com/c/product/1434274-REG/lectrosonics_spdr_stereo_portable_digital.html?gclid=CjwKCAjw1YCkBhAOEiwA5aN4AbtcsUbu8gDVaxmuWeYg3A_R0eGMZo1b_2m1VkaJ-4eFYOzbi9TQYhoC0gsQAvD_BwE
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the DPA D:VICE is so small it fits into my coin compartment of my wallet - nobody will ever check that. super easy so smuggle in.
also you do not need to hide your phone, it raises no suspicion when looking at it and the phones battery lasts for hours for recording 24 bit.
here you will have an extra device with alcaline batteries (that can be connected remotely to - your phone).
unless the recording quality isnt muuuuch better than the DPA D:VICE i don't think this is much of a progress for stealth taping?
This recorder is 32 bit with 5v plug in power. I could plug my 4061s straight in and easily have the entire rig self-contained with no wires to show and no real need to check levels.
Sounds like a win to me.
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You can also use the new version of the Sonosax sx-m2d2 "electret".
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Competition:
- https://lectrosonics.com/spdr-stereo-personal-digital-recorder.html
- Tentacle: not yet
32bit ?
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You can also use the new version of the Sonosax sx-m2d2 "electret".
new version ?
i love my current
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Does anyone know what the "locking" 3.5mm stereo mic / stereo line input means?
Also, I don't know what 24 or 32 bit "float" means, although I imagine it's discussed somewhere on this board - any brief explanation or link would be appreciated.
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Does anyone know what the "locking" 3.5mm stereo mic / stereo line input means?
Also, I don't know what 24 or 32 bit "float" means, although I imagine it's discussed somewhere on this board - any brief explanation or link would be appreciated.
it means the jack is threaded so you can screw down on it if you have that locking nut
24 bit v 32 bit float are different bit depths in recording. 32bit float is newer and can give you more room. thats a quick and dirty. there is a dedicated thread for it if i recall
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Does anyone know what the "locking" 3.5mm stereo mic / stereo line input means?
Also, I don't know what 24 or 32 bit "float" means, although I imagine it's discussed somewhere on this board - any brief explanation or link would be appreciated.
it means the jack is threaded so you can screw down on it if you have that locking nut
24 bit v 32 bit float are different bit depths in recording. 32bit float is newer and can give you more room. thats a quick and dirty. there is a dedicated thread for it if i recall
So you think any microphone with an 1/8" male plug could be locked in so it wouldn't accidentally come out when in use? If so, that would be a very good feature, but I've never heard of such a thing.
I know what 24 and 32 bit mean, just had never heard of this "float" until very recently and don't know what that really means.
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So you think any microphone with an 1/8" male plug could be locked in so it wouldn't accidentally come out when in use? If so, that would be a very good feature, but I've never heard of such a thing.
No, the plug has to be threaded as well. They're mostly common in wireless beltpacks. I'm not even sure if the different manufacturers all use the same one.
(https://cdn.sajelectronics.com/file/thebroadcastshop-3/images/mfr/Rode/large/micon1.jpg)
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"I know what 24 and 32 bit mean, just had never heard of this "float" until very recently and don't know what that really means."
It means you never have to set gain levels again.
"To put it in perspective, 16-bit audio is capable of recording sound with a dynamic range of up to 96.3 decibels. 24-bit audio recordings can capture a dynamic range of up to 144.5 dB. Meanwhile, 32-bit float audio can capture the absolutely ludicrous range of up to 1,528 dB. That’s not only massively beyond the scope of 24-bit audio, but it’s beyond the scale of what even counts as a sound on Earth." https://www.wired.com/story/32-bit-float-audio-explained/
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is the electret sx-m2d2 option an either / or with phantom or can you switch back and forth easily?
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ask mr.SAX
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"I know what 24 and 32 bit mean, just had never heard of this "float" until very recently and don't know what that really means."
It means you never have to set gain levels again.
"To put it in perspective, 16-bit audio is capable of recording sound with a dynamic range of up to 96.3 decibels. 24-bit audio recordings can capture a dynamic range of up to 144.5 dB. Meanwhile, 32-bit float audio can capture the absolutely ludicrous range of up to 1,528 dB. That’s not only massively beyond the scope of 24-bit audio, but it’s beyond the scale of what even counts as a sound on Earth." https://www.wired.com/story/32-bit-float-audio-explained/
32-bit float is indeed a useful improvement but not the massive savior it is made out to be. 1528 dB of digital resolution is really not "massively beyond the scope of 24-bit audio" which has 144dB of resolution, as there does not exist an analog source with more than about 130dB of dynamic range, in theory. In reality, any live concert recording, even a SBD, has a dynamic range of around 100 dB at best, so the advantages over the common 24-bit interfaces, with levels set conservatively, are limited. Audience recordings are more like 70 dB dynamic range with the ambient crowd noise
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"I know what 24 and 32 bit mean, just had never heard of this "float" until very recently and don't know what that really means."
It means you never have to set gain levels again.
"To put it in perspective, 16-bit audio is capable of recording sound with a dynamic range of up to 96.3 decibels. 24-bit audio recordings can capture a dynamic range of up to 144.5 dB. Meanwhile, 32-bit float audio can capture the absolutely ludicrous range of up to 1,528 dB. That’s not only massively beyond the scope of 24-bit audio, but it’s beyond the scale of what even counts as a sound on Earth." https://www.wired.com/story/32-bit-float-audio-explained/
32-bit float is indeed a useful improvement but not the massive savior it is made out to be. 1528 dB of digital resolution is really not "massively beyond the scope of 24-bit audio" which has 144dB of resolution, as there does not exist an analog source with more than about 130dB of dynamic range, in theory. In reality, any live concert recording, even a SBD, has a dynamic range of around 100 dB at best, so the advantages over the common 24-bit interfaces, with levels set conservatively, are limited. Audience recordings are more like 70 dB dynamic range with the ambient crowd noise
True that, but my F3 has already saved my bacon twice when the music took a sudden and dramatic increase in volume, as opposed to the case with the Sony PCM M10 that I was using on a different rig. The F3 also means that I do not even have to think about what a conservative setting might be at a show of improvised music with a wide dynamic range in a small room. Live recording is stressful enough, so even limited improvements in the number of things to think about can be welcome.
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"I know what 24 and 32 bit mean, just had never heard of this "float" until very recently and don't know what that really means."
It means you never have to set gain levels again.
"To put it in perspective, 16-bit audio is capable of recording sound with a dynamic range of up to 96.3 decibels. 24-bit audio recordings can capture a dynamic range of up to 144.5 dB. Meanwhile, 32-bit float audio can capture the absolutely ludicrous range of up to 1,528 dB. That’s not only massively beyond the scope of 24-bit audio, but it’s beyond the scale of what even counts as a sound on Earth." https://www.wired.com/story/32-bit-float-audio-explained/
32-bit float is indeed a useful improvement but not the massive savior it is made out to be. 1528 dB of digital resolution is really not "massively beyond the scope of 24-bit audio" which has 144dB of resolution, as there does not exist an analog source with more than about 130dB of dynamic range, in theory. In reality, any live concert recording, even a SBD, has a dynamic range of around 100 dB at best, so the advantages over the common 24-bit interfaces, with levels set conservatively, are limited. Audience recordings are more like 70 dB dynamic range with the ambient crowd noise
This misses part of the point. For me, the advantages of 32Bit Float are most noticeable when the range of a recorded piece is dramatic. I was recording a show last week where the band abandoned the stage and mics and did three songs busking in the audience with no amplification. The first song farthest away from me, I could not even hear. Each song got closer to where i was recording, but for this segment the band was a violin, tambourine and vocal with no amplification. Some storytelling too, which was just impossible to hear. I boosted that part of the show over 20dB and there was no added noise at all. The wave form went from virtually a straight line to full sized, and the sound was superb. Music sweet and vocals clear. It was amazing. I know there are lots who say 32Bit float is unnecessary, but it comes in handy and does no harm. I do not think it is being hails as a massive savior. It is just better than 24Bit as 24 was better than 16.
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Concur 100 percent with the above
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32 bit is a godsend when you're not in a position where you can check your levels at all
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I totaly agree with Dallman.
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Without getting too technical.. with 32bit-float recording the real-world dynamic range of the recording is determined by either the limits of the microphones or the the analog input stage of the recorder, whichever is less. Although neither of those exceed the ~144dB dynamic range of the 24bit-fixed file format*, the absence of any recording trim controls in 32bit-float mode eliminates the possibility of error in setting initial input trim.
In reality, when recording content of low or typical loudness it shields folks from worrying about recording at what would otherwise seem very low input trim settings where the the meters are barely registering. The important thing is not really the 32bit file format itself (24bits is more than enough) but a good enough analog input stage and the design of the ADC in the recorder.
Using 32bit-float will require level setting (normalization) and file format conversion afterward, but eliminates level setting in the heat of the moment, be it necessary or not.
Perhaps ironically, it actually prevents tapers from "running levels hot", as many have historically preferred for reasons good or otherwise. Using a recorder in 32-bit mode, "running hot" can only be achieved within an external preamp.
*A quick search indicates that real-world dynamic range capability of the SD Mixpre recorders is around 142db, the Zoom F series around 131db.
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*A quick search indicates that real-world dynamic range capability of the SD Mixpre recorders is around 142db, the Zoom F series around 131db.
by standard measurement 142 dB is impossible. the standard 150 kohm input noise measurements yields a maximum value of 131 dBu/133 dBv, as even sound devices acknowledges
https://www.sounddevices.com/microphone-preamp-noise/
"EIN is helpful as it removes gain from the equation and makes apples-to-apples comparisons easier. EIN is expressed in dBV or dBu. The lower the number, the better the EIN. This number is properly measured using 150 ohms as an input terminator. The very best EIN that can be achieved is -133 dBV, since this is noise purely from a 150 ohm resistor."
Perhaps there is some yet-unknown input source with greater than 142 dB of range, that they are using to derive this number, or they somehow have a dual-ranging switching analog circuitry similar to the multi-range ADCs
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Thanks for that clarification, which further emphasizes the basic point.
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I wonder if a lesson to be learned from this discussion is that even when recording in 24 bit some of us would be surprised how low we can get away with setting our levels.
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Try 12db peaks at a noisy venue for 24 bit.
In fact, I'll do that tonight...
Is anyone still pulling at 96kHz or 192?
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Try 12db peaks at a noisy venue for 24 bit.
In fact, I'll do that tonight...
Is anyone still pulling at 96kHz or 192?
^^I shoot for -18dB peaks running a MixPre-6 (V1) in 24bit/192. I'm recording classical/opera. Sometimes someone will decide to get a lot louder, and the ambient noisefloor is significantly higher than the noisefloor of the MixPre at -18 after I bring it up in post, so no cost to having the extra safety buffer as far as I can tell. To achieve this, I'm giving most mics around 20-25dB of gain, which happens to be right around where we maximize S/N ratio on most recorders according to the Schoeps talks this year anyways.
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For me, it's chiefly a matter of not needing to set levels, although in the old days I used to enjoy getting to the end of a classical music recording with the a peak level of 0.5dB. The brinkmanship would keep me awake during the more tedious recitals. I am still awaiting the Zoom M3 ordered months ago, which doesn't even bother with a display of any kind. Just a power button and a record button, and that's it (more or less). Welcome to the future.
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Here's a plus for 32bit based on my >:D experience last night. I forgot to lock the recorder (PCM-A10) and when I checked my levels via phone 10 mins into the first act, levels were maxed and I was hard brickwalling.
Had I been using a 32 bit recorder (like the Deity or F3), this would been recoverable. Luckily I only lost 10 minutes and none of the headliner.
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For me, it's chiefly a matter of not needing to set levels, although in the old days I used to enjoy getting to the end of a classical music recording with the a peak level of 0.5dB. The brinkmanship would keep me awake during the more tedious recitals. I am still awaiting the Zoom M3 ordered months ago, which doesn't even bother with a display of any kind. Just a power button and a record button, and that's it (more or less). Welcome to the future.
I googled Zoom M3 and I just got a microphone. Any info about the recorder?
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I googled Zoom M3 and I just got a microphone. Any info about the recorder?
It's Zoom's camera-mounted shotgun mic that has its own builtin 32 bit recorder and M/S decoder. I expect the recorder may be similar to the F3, though I haven't compared specs, except that M3 is a self-contained mic + recorder system/not for external mics. Haven't tried it, but it has to be way better than those garbage rode mini shotguns I see on everyone's camera around LA.
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I googled Zoom M3 and I just got a microphone. Any info about the recorder?
It's Zoom's camera-mounted shotgun mic that has its own builtin 32 bit recorder and M/S decoder. I expect the recorder may be similar to the F3, though I haven't compared specs, except that M3 is a self-contained mic + recorder system/not for external mics. Haven't tried it, but it has to be way better than those garbage rode mini shotguns I see on everyone's camera around LA.
The recorder part might be great, but I don't think we have many data points to judge the quality of Zoom's mic designs other than the modular models made for the H series, and we can't really test the quality of those independent of the recorders they attach to with their proprietary connection. Rode makes some excellent mics at their higher price tiers, but his M3 might be no better than the cheap Rode mini shotguns. For either product, they probably aren't marketing to the same people who might buy a Sanken CMS-50.
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Good news, D..
Here's a plus for 32bit based on my >:D experience last night. I forgot to lock the recorder (PCM-A10) and when I checked my levels via phone 10 mins into the first act, levels were maxed and I was hard brickwalling.
Had I been using a 32 bit recorder (like the Deity or F3), this would been recoverable. Luckily I only lost 10 minutes and none of the headliner.
True that if you had been using a 32 bit recorder, the input level would have been out of your control and low enough from the start (extremely low). However, you can run the A10 in essentially the same way by resisting the temptation to increase recording levels above whatever minimum setting is needed to avoid overload in the loudest recording situations you encounter.
Doing so will not be a problem as long as..
[snip..] the ambient noisefloor is significantly higher than the noisefloor of the MixPre at -18 [of your recording chain, in this case determined by microphone self-noise>recorder EIN] after I bring it up in post, so no cost to having the extra safety buffer [..snip]
SMsound describes classical/opera which has an ambient noise floor that is about as quiet as any taper-recorded live performance ever gets. The biggest problem for tapers is that not increasing recording level is a very difficult temptation to overcome! it's one that has become ingrained by habit as it used to be important but generally isn't any longer, 32bit or not.
The parallel switching ADC designs and very low EIN of 32bit float recorders help extend dynamic range sufficiently to fully accommodate very low ambient noisefloors that don't occur in concert taper situations. Most modern recorders that are not 32bit float can be used in essentially the same way as 32bit recorders for concert recording. I use a DR2d for classical recording, which when recording in 24bit probably has a real-world dynamic range of only around 18bit equivalent or so at best. I've never actually measured it. But as they should be, the noisefloor of the recordings I make with it are dominated by the ambient noisefloor of the hall. Even with an effective range of only about 18bit and the DR2d's input levels remaining the same all the time, I use only two different gain settings on the preamp upstream of the recorder- one for classical recording (determined by the need to keep the recording chain noisefloor lower than the ambient noisefloor) and one for very much louder PA amplified stuff (determined by the needed to keep the loudest SPL from clipping). Once those settings have been determined, there is no need to look at or worry about them again.
Using the Zoom F8 instead and recording in 24bit, I don't even need two gain settings.
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Good news, D..
Here's a plus for 32bit based on my >:D experience last night. I forgot to lock the recorder (PCM-A10) and when I checked my levels via phone 10 mins into the first act, levels were maxed and I was hard brickwalling.
Had I been using a 32 bit recorder (like the Deity or F3), this would been recoverable. Luckily I only lost 10 minutes and none of the headliner.
True that if you had been using a 32 bit recorder, the input level would have been out of your control and low enough from the start (extremely low). However, you can run the A10 in essentially the same way by resisting the temptation to increase recording levels above whatever minimum setting is needed to avoid overload in the loudest recording situations you encounter.
Doing so will not be a problem as long as..
[snip..] the ambient noisefloor is significantly higher than the noisefloor of the MixPre at -18 [of your recording chain, in this case determined by microphone self-noise>recorder EIN] after I bring it up in post, so no cost to having the extra safety buffer [..snip]
SMsound describes classical/opera which has an ambient noise floor that is about as quiet as any taper-recorded live performance ever gets. The biggest problem for tapers is that not increasing recording level is a very difficult temptation to overcome! it's one that has become ingrained by habit as it used to be important but generally isn't any longer, 32bit or not.
The parallel switching ADC designs and very low EIN of 32bit float recorders help extend dynamic range sufficiently to fully accommodate very low ambient noisefloors that don't occur in concert taper situations. Most modern recorders that are not 32bit float can be used in essentially the same way as 32bit recorders for concert recording. I use a DR2d for classical recording, which when recording in 24bit probably has a real-world dynamic range of only around 18bit equivalent or so at best. I've never actually measured it. But as they should be, the noisefloor of the recordings I make with it are dominated by the ambient noisefloor of the hall. Even with an effective range of only about 18bit and the DR2d's input levels remaining the same all the time, I use only two different gain settings on the preamp upstream of the recorder- one for classical recording (determined by the need to keep the recording chain noisefloor lower than the ambient noisefloor) and one for very much louder PA amplified stuff (determined by the needed to keep the loudest SPL from clipping). Once those settings have been determined, there is no need to look at or worry about them again.
Using the Zoom F8 instead and recording in 24bit, I don't even need two gain settings.
Completely agree, the issue was that the unlocked recorder was bumping against other things in my pocket, which was the cause for increased levels. I need things to be as foolproof as possible!
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I hear that! Practicality reins supreme when taping in real world situations.
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I tend to set my levels on the conservative side, and when I use my dr2d, even with the gain jacked up to the max, I don't come anywhere near peaking even if I tried. I record in 24 bit and haven't ever noticed any noise issues due to the conservative levels. (I have, of course, gotten lots of noise from the screaming hooting whistling screeching cackling talking buffoons that always seem to surround me). I don't really need any new recorders but I'm interested in seeing the price point on this one, as the size looks like a dream come true.
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I've a general question which applies to a potential application of these recorders in particular. Can the timecode capability of these types of recorders sync several of them accurately enough to avoid having to do the align + stretch/shrink thing in post? I'm not well versed in time-code, but I do know that the accuracy required to sync audio being recorded across multiple machines with phase-coherent accuracy is far far greater than what is required to sync visual frame rates (the difference in frame rate and sample rate is something like two orders of magnitude). Timecode is not wordclock.
In a perfect word I'd use a single recorder similar to this which features 6 or 8 input channels. Since that's a unicorn wish, I next wonder about running three or four of these recorders sync'd together and controlled by the phone app. Possible? Realistic? Phase accurate?
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I've a general question which applies to a potential application of these recorders in particular. Can the timecode capability of these types of recorders sync several of them accurately enough to avoid having to do the align + stretch/shrink thing in post? I'm not well versed in time-code, but I do know that the accuracy required to sync audio being recorded across multiple machines with phase-coherent accuracy is far far greater than what is required to sync visual frame rates (the difference in frame rate and sample rate is something like two orders of magnitude). Timecode is not wordclock.
In a perfect word I'd use a single recorder similar to this which features 6 or 8 input channels. Since that's a unicorn wish, I next wonder about running three or four of these recorders sync'd together and controlled by the phone app. Possible? Realistic? Phase accurate?
timecode isnt really useful for what we do, all of the recorders are still running on their own clocks
the only thing you are buying with timecode is making the (not-too-hard) process of finding whatever starting sample you align multiple wavs with slightly easier, but they will still drift apart in time, a little, or a lot, depending on gear
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As I suspected, thanks.
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I googled Zoom M3 and I just got a microphone. Any info about the recorder?
It's Zoom's camera-mounted shotgun mic that has its own builtin 32 bit recorder and M/S decoder. I expect the recorder may be similar to the F3, though I haven't compared specs, except that M3 is a self-contained mic + recorder system/not for external mics. Haven't tried it, but it has to be way better than those garbage rode mini shotguns I see on everyone's camera around LA.
The recorder part might be great, but I don't think we have many data points to judge the quality of Zoom's mic designs other than the modular models made for the H series, and we can't really test the quality of those independent of the recorders they attach to with their proprietary connection. Rode makes some excellent mics at their higher price tiers, but his M3 might be no better than the cheap Rode mini shotguns. For either product, they probably aren't marketing to the same people who might buy a Sanken CMS-50.
I think there's a dedicated thread somewhere for this device, but as far as I am aware, the only music test from it online is this - https://youtu.be/4A3S1tuq2GQ - where the stereo width indicated on screen is clearly not what is being heard, but I think that's an error in making the video. Anyway, while it's hard to know what this particular pipe organ should sound like, I would say that to my ears the sound is no disaster, especially at the modest price. And now, back to the Deity..
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any idea of its release date on the market ?
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any idea of its release date on the market ?
I asked a German vendor about the release date...his answer was August/September...price unknown!
https://www.marcotec-shop.de/de/deity-pr-2-pocket-audio-recorder.html
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any idea of its release date on the market ?
I asked a German vendor about the release date...his answer was August/September...price unknown!
https://www.marcotec-shop.de/de/deity-pr-2-pocket-audio-recorder.html
Gotham Sound says September
https://www.gothamsound.com/product/pr-2-pocket-recorder
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oh boy sooo looking forward to getttng a DR2
thanks y'all for reporting on release date... etc..
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What are the best mics you could power with this unit?
Spec's say:
3V / 5V / LINE Switchable mic power
I think 5V is just under what you'd ideally want for DPA 4060/4061... Maybe the low voltage version of those?
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What are the best mics you could power with this unit?
Spec's say:
3V / 5V / LINE Switchable mic power
I think 5V is just under what you'd ideally want for DPA 4060/4061... Maybe the low voltage version of those?
I am 99% sure 4060 & 4061 can be fully powered with 5-10v. 4063 only needs 3v.
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seems you can control the units with this software
https://www.sidus.link/sidusAudio/software
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What are the best mics you could power with this unit?
Spec's say:
3V / 5V / LINE Switchable mic power
I think 5V is just under what you'd ideally want for DPA 4060/4061... Maybe the low voltage version of those?
I am 99% sure 4060 & 4061 can be fully powered with 5-10v. 4063 only needs 3v.
I used the 4060s with the plug in power of the PMD620 at just under 5V and no probs at loud shows or otherwise.
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this paired with an OG pre-amp would make a cool rig and small footprint and you could stealth it as well
i could also use it as a backup on my m2d2 or primary if i prefer the adac
m2d2>pr-2 would be a really small rig
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Still waiting for a date or price announcement on this one. And an assessment of the preamp/AD virtues.
The stereo input is selectable for line-in or 3V or 5V pip. Anyone know if the dpa 4060 etc. will run with 5V and this sort of adapter:
https://www.ebay.com/itm/275465573952?chn=ps&mkevt=1&mkcid=28&srsltid=AfmBOopwvh2IUurEIoj7ed6-zo_Yv_GA0cIJLRKCsuZDvKEV6IkKpZtg7oc (https://www.ebay.com/itm/275465573952?chn=ps&mkevt=1&mkcid=28&srsltid=AfmBOopwvh2IUurEIoj7ed6-zo_Yv_GA0cIJLRKCsuZDvKEV6IkKpZtg7oc)
Jeff
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Answering my own question: I saw online that someone had measured the PiP open circuit voltage of the Sony D100 at 4.75 volts (most Tascams seem well under 3V), put new batteries in my old D100 and got a pretty good signal from 4060s with this cable. No recording made to test for noise or other problems, but the response seemed normal enough that I may try the Deity PR-2 when it lands, if the price is okay.
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Apparently, it's going to be released in September 2023:
https://www.gothamsound.com/product/pr-2-pocket-recorder (https://www.gothamsound.com/product/pr-2-pocket-recorder)
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The stereo input is selectable for line-in or 3V or 5V pip. Anyone know if the dpa 4060 etc. will run with 5V and this sort of adapter:
https://www.ebay.com/itm/275465573952?chn=ps&mkevt=1&mkcid=28&srsltid=AfmBOopwvh2IUurEIoj7ed6-zo_Yv_GA0cIJLRKCsuZDvKEV6IkKpZtg7oc (https://www.ebay.com/itm/275465573952?chn=ps&mkevt=1&mkcid=28&srsltid=AfmBOopwvh2IUurEIoj7ed6-zo_Yv_GA0cIJLRKCsuZDvKEV6IkKpZtg7oc)
Should do fine. 5V is what DPA specs as nominal powering voltage for the miniature mics (4063 being the exception).
That cable looks like the ones many of us here have hacked together in the past. Assuming its wired with both shield/grounds connnected to sleeve, Left center conductor to tip, Right center conductor to ring.
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is it diety 2? or zoom F3 - how bout some pro cons - d2 vs f3 talks.....???
looks like diety less metal than zoom and more compact? to get in venues stealth
if you don't need xlr
everyones thoughts?
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Looking at the specs, dynamic range is 123 dB. That figure makes me wonder if it really is an autoranging multi-ADC setup like Tascam, Zoom, Sound Devices, and Stagetec employ. Zoom F6 dynamic range is 132 dB using 2 ADCs; SD MixPre-II series is 142 dB using 3 ADCs.
It looks like it can only accept timecode from other Diety units, but an interesting bit from the Gotham video is that the PR-2 can be a timecode source via 3.5 mm TRS output. In theory, than means you could slave or jam sync it to units from other manufacturers, right? The F6 does timecode via 3.5 mm so it would seem like a perfect partner.
Regarding powering DPA lavs: the 406x are supposed to get 8V, so I wonder how compromised their performance is with only 5V. You'd probably be OK with loud sources if you use the 4061, but for what I record I would be concerned about increased self-noise and reduced dynamic range.
Why can't someone make a little recorder like this with PIP settings from 3-9 V (or more)? Sonosax had this figured out many years ago with the miniR82.
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is it diety 2? or zoom F3 - how bout some pro cons - d2 vs f3 talks.....???
looks like diety less metal than zoom and more compact? to get in venues stealth
if you don't need xlr
everyones thoughts?
The F3 is proven to be a great unit, sounds identical to its F6 and F8 brethren, can power any full-size mic, and is cheap. It's very well built aside from the ridiculous power/REC switch. It needs an external power bank if using P48 mics for any significant amount of time.
The FR-2 will most likely have better industrial design and higher quality materials, and probably a bit more expensive. Much more compact. You'll also be limited in what mics you can use it with without an external preamp or mic battery box.
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Regarding powering DPA lavs: the 406x are supposed to get 8V, so I wonder how compromised their performance is with only 5V. You'd probably be OK with loud sources if you use the 4061, but for what I record I would be concerned about increased self-noise and reduced dynamic range.
Why can't someone make a little recorder like this with PIP settings from 3-9 V (or more)? Sonosax had this figured out many years ago with the miniR82.
?????
In the parallel universe where I ran a miniR82 for many years, the unit's eight tracks included a maximum of four analogue channels: a line-in on a 3.5mm stereo plug and a mic-in which could be wired for line-in, phantom power or (ONLY) 3V PiP. Sonosax provided schematics how to wire for phantom or COS11 or DPA4063 mics (both run on 3V), and they sold me a DPA4063 set wired correctly (I think they also sold the Sankens wired). Here is the schematic:
https://www.sonosax.ch/wp-content/uploads/2016/12/MINIR82-MIC-WIRING.pdf (https://www.sonosax.ch/wp-content/uploads/2016/12/MINIR82-MIC-WIRING.pdf)
The later M2D2 preamp/AD originally had no plug-in power options, phantom only, but after a few years I understand they offered an updated version with some sort of PiP, I think it was DPA4060 etc. friendly, but I can't find any info on this update anymore on their website, does anyone know if it's still available and what the mic specs are?
Jeff
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thanks for feedback guys
whos getting a diety?
looks like i/m going that way
Taz
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If it's reasonably priced I'll likely try it, in spite of the 48 kHz limit (I usually tape at 96 kHz). I will also be testing if the cable I linked above will power DPA4011 or 4015 caps with the active cable/preamp (these all work with the now deceased MMA-A and iPhone powering). If it's expensive I'll wait for the rave reviews before biting.
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Regarding powering DPA lavs: the 406x are supposed to get 8V, so I wonder how compromised their performance is with only 5V. You'd probably be OK with loud sources if you use the 4061, but for what I record I would be concerned about increased self-noise and reduced dynamic range.
Why can't someone make a little recorder like this with PIP settings from 3-9 V (or more)? Sonosax had this figured out many years ago with the miniR82.
?????
In the parallel universe where I ran a miniR82 for many years, the unit's eight tracks included a maximum of four analogue channels: a line-in on a 3.5mm stereo plug and a mic-in which could be wired for line-in, phantom power or (ONLY) 3V PiP. Sonosax provided schematics how to wire for phantom or COS11 or DPA4063 mics (both run on 3V), and they sold me a DPA4063 set wired correctly (I think they also sold the Sankens wired). Here is the schematic:
https://www.sonosax.ch/wp-content/uploads/2016/12/MINIR82-MIC-WIRING.pdf (https://www.sonosax.ch/wp-content/uploads/2016/12/MINIR82-MIC-WIRING.pdf)
The later M2D2 preamp/AD originally had no plug-in power options, phantom only, but after a few years I understand they offered an updated version with some sort of PiP, I think it was DPA4060 etc. friendly, but I can't find any info on this update anymore on their website, does anyone know if it's still available and what the mic specs are?
Jeff
I was being pretty careless in my response there. I was really thinking of the R82 being a very compact unit that could power phantom mics. I really had no idea what it could do for PIP. Thanks for the details, though.
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
"needed" is a loaded word. The Berlin Philharmonic records their live stuff at 192 kHz, or maybe even 384. I have tried 192 (and DSD), and decided 96 kHz is fine for me, including editing with Rx. Some people who do 48 kHz think 44.1 is undesirable. Quality of AD is not a 1-1 matchup with sampling frequency, either. So I'm curious about how the Deity implementation of 48 k sounds.
Jeff
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Just getting used to the F3 for my low security gigs with NBox Platinum but curious about the Deity until. For now my high >:D rig is babynbox dr-2d schoeps MK* but open to reports. Who says I can't have both.
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
The maths is pretty clear. The Nyquist–Shannon sampling theorem shows that you are able to perfectly reconstruct an analogue sound wave if the sampling rate of the digital recording is at least twice that of the original wave. This means that a 44.1 kHz can perfectly reproduce all frequency content below 22.05 kHz. When I was a teenager, I could only hear up to 20 kHz or so. At 39, this is down to 17 kHz. Your microphone's frequency response often won't be higher than 20 kHz, and if it is, your headphones' often won't be. I can see why you'd want to record at 48 kHz if you are dealing with video (it's the default for DVD and other video content), but CD quality (44.1 kHz) is not only 'good enough' - it reproduces any sound we can hear perfectly. I'd understand the use of higher sampling rates for scientific purposes, but not for listening.
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
The maths is pretty clear. The Nyquist–Shannon sampling theorem shows that you are able to perfectly reconstruct an analogue sound wave if the sampling rate of the digital recording is at least twice that of the original wave. This means that a 44.1 kHz can perfectly reproduce all frequency content below 22.05 kHz. When I was a teenager, I could only hear up to 20 kHz or so. At 39, this is down to 17 kHz. Your microphone's frequency response often won't be higher than 20 kHz, and if it is, your headphones' often won't be. I can see why you'd want to record at 48 kHz if you are dealing with video (it's the default for DVD and other video content), but CD quality (44.1 kHz) is not only 'good enough' - it reproduces any sound we can hear perfectly. I'd understand the use of higher sampling rates for scientific purposes, but not for listening.
:coolguy: Plainspeak true dat
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
The maths is pretty clear. The Nyquist–Shannon sampling theorem shows that you are able to perfectly reconstruct an analogue sound wave if the sampling rate of the digital recording is at least twice that of the original wave. This means that a 44.1 kHz can perfectly reproduce all frequency content below 22.05 kHz. When I was a teenager, I could only hear up to 20 kHz or so. At 39, this is down to 17 kHz. Your microphone's frequency response often won't be higher than 20 kHz, and if it is, your headphones' often won't be. I can see why you'd want to record at 48 kHz if you are dealing with video (it's the default for DVD and other video content), but CD quality (44.1 kHz) is not only 'good enough' - it reproduces any sound we can hear perfectly. I'd understand the use of higher sampling rates for scientific purposes, but not for listening.
I still record at 44.1 kHz. I haven't found any noticeable difference using any other sampling frequency that results in better quality for what I do. I've tried comparisons using up to 96kHz and it just doesn't add anything to what I'm doing except for larger file size. This is for concert recordings that I'm doing as a hobby on my own dime.
When I've had paying gigs I've done all my recordings in 96 kHz sampling freq. More for the client's expectations than anything else.
Interested in this recorder. My small recorder is the Korg MR1 and while it sounds great and is fairly reliable I've had to custom modify it and it is getting a little long in the tooth.
I like the idea of having a recorder that doesn't have any internal mics.
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Record at whatever sample rate you like at or above 44.1kHz. I don't dismiss folks who claim higher sample rates makes a significant difference to them. Justified or not, client expectations fall into that category. Its just that plenty of other things have greater implications.
It is fun to record bats and bugs or whatever at high rate and playback at a lower rate to make the ultrasonic singing audible, but that's a different thing than the live music focus here at TS. In regard to music recording, I have experienced higher sample rates sounding somewhat different in some instances, yet suspect that's due to the quality of specific gear, the specific ADC/DAC implementations and surrounding circuitry, rather than something consistent enough to form the basis of a blanket rule I'd feel comfortable relying upon without that kind of careful listening. So on that cost/benefit basis I record at 24/48kHz as a way of keeping file storage size reasonable - it's consistent with both "the maths" and my listening experiments.
The strongest argument I've heard for higher sample rates for music recording is to provide increased calculation bandwidth during processing. But processing can be done at a higher rate than the native depth of the source, in addition to being performed in a calculation space of greater bit-depth than that of the native files (which is how all modern DAW software operates). Just use that 2X, 4X, 8X, or AUTO "up-sample upon render" option in your plugins if they offer it when using a machine capable of the additional processing overhead.
There is actually a rather strong argument for limiting audio bandwidth so as not to exceed the limits of human high-frequency audibility in an excessive way first, prior to processing at a multiple of the sample-rate, which may be an unexpected benefit of the lower band-pass filtering inherent to 48 and 44.1kHz.
I like the simple function and form factor of this recorder and am quite interested in it, but need more than two channels so will likely continue using DR2d.
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do u folks think 96khz is needed?
i usually tape in 48 on a sony a19
Taz
The maths is pretty clear. The Nyquist–Shannon sampling theorem shows that you are able to perfectly reconstruct an analogue sound wave if the sampling rate of the digital recording is at least twice that of the original wave. This means that a 44.1 kHz can perfectly reproduce all frequency content below 22.05 kHz. When I was a teenager, I could only hear up to 20 kHz or so. At 39, this is down to 17 kHz. Your microphone's frequency response often won't be higher than 20 kHz, and if it is, your headphones' often won't be. I can see why you'd want to record at 48 kHz if you are dealing with video (it's the default for DVD and other video content), but CD quality (44.1 kHz) is not only 'good enough' - it reproduces any sound we can hear perfectly. I'd understand the use of higher sampling rates for scientific purposes, but not for listening.
Also, good luck finding a PA that approaches 20K. Many mics don't go out 20k as well, especially when directional components are considered. I think people who grew up with equalizers over-value the significance of even 16K, as the EQ acts as a wide band and influences frequencies below it
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Just getting used to the F3 for my low security gigs with NBox Platinum but curious about the Deity until. For now my high >:D rig is babynbox dr-2d schoeps MK* but open to reports. Who says I can't have both.
You can have both (all three really), but they are very different animals. Not sure what a "low security" gig is (I assume that's still >:D but not 'open taping'? A gig you still have to sneak gear into?) Since the F3 has phantom power the N-Box platinum solution is quite a bit of extra gear over just using the off the shelf solutions like a CMC1L or CCM, which would eliminate the N-Box and extra cabling. Putting aside 32-bit for a moment, depending on your situation, using an F3, over a smaller cheaper common 2-channel recorder like an DR-2D ( or A10,M10, R07, etc) the advantage is not really clear. I'm not sure it outperforms the other recorders significantly enough on unbalanced line in to justify the extra gear. 32bit aside, F3 is certainly not the only one in its class.Centrance Mixerface comes to mind and is about the same size as the F3, The TASCAM DR40 is only slightly larger (and is slimmer and nearly entirely plastic), with the DR100 series stepping up from that in size a bit
Back to the subject of the thread, the Deity PR2 is really nothing like phantom recorders or the smaller line-in recorders. The former is larger and offers mic powering you may not need, the latter cant really power mics at all without additional equipment.
The PR2 looks to be the size of an A10 based on the pictures (size and weight not on product page but you can gauge by the size of the AA batteries). Timecode, 32-bit recording, and the ability to power mics at 5V in the form factor of one of the smallest 2-ch line in recorders is it advantage. First advantage is essentially useless to tapers, the second may be useful (especially given the level display on the pr2 appears to be as crippled as the TRX) but not really make-or-break, but the ability to properly power two mics at 5V in that form factor is a game changer. Nothing exists like that now. you need to step up to $1000+ solution to find that from Zaxcom, Lectrosonics, Sonosax, etc.... and most are larger and metal. the tentacle track-e comes close, but is only mono and youd need two devices at $600 and then have to sync them in post. Real 5V PIP power opens up proper powering at high-SPL of all lav mics, as well as the higher end DPA series, so more of a breakthrough for DPA guys. for Schoeps guys it seems to be the equivalent of just another A10 with 32-bit float replacing decent level meters.
In reality, implementation matters more than feature set. id rather have a reliable easy-to-use recorder with a battery box than an all-in-one that doesn't bring home the goods every time. So as you said.. i will wait for reports!
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hows all this compare to
the Tascam dr10-L?
is this unit up to par with diety and zoomf3?
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hows all this compare to
the Tascam dr10-L?
is this unit up to par with diety and zoomf3?
Different animals
DR10-L is one channel, and cannot power either phantom mics or lav mics (in our application)
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Different animals
DR10-L is one channel, and cannot power either phantom mics or lav mics (in our application)
------
ok thx u
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Different animals
DR10-L is one channel, and cannot power either phantom mics or lav mics (in our application)
------
ok thx u
There is a quote button in the upper right of each post. ;)
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rairun, what you say is entirely correct with regard to sampling and reconstruction per se. However, any significant signal energy at or above 1/2 the sampling frequency will cause aliasing distortion, and thus we are all sternly commanded to filter out such energy prior to (or as part of) the conversion process.
Significant energy at such frequencies is rare in real-world sound, as opposed to specially generated test signals designed to challenge a recording system. But in order to handle the worst cases without audible aliasing, digital audio systems conventionally use low-pass filtering with stopband attenuation of 60 - 80 dB or even more. This forces those filters to be awfully steep at sampling rates when there isn't much margin between their turnover point and the Nyquist limit (e.g. the narrow interval between 20 kHz and 22.05 kHz in the case of 44.1 kHz sampling).
The anti-aliasing filters in the PCM-F1 were 9th-order IIRC, and the ones in the PCM-1600 and 1610 were 11th or even 13th. That's way more than would almost ever be needed in my opinion, but it's been the general practice for decades. Such filters can have audible effects on impulse response. They can do things that have no counterpart in the real world of sound, such as ringing that starts before an impulse has actually begun (as well as continuing after the sound has stopped, as one might expect).
At higher sampling rates, on the other hand, the filters don't need to be nearly as steep--and even if they are, with their turnover point an octave higher there will still be far less time-domain nonsense below 20 kHz. Thus the impulse response of a system with a higher sampling frequency can be better--even (occasionally) audibly so under certain conditions. That said, you are also perfectly right about the limitations of most playback systems--especially conventional dynamic loudspeakers. Most of the time when people have provably, repeatably heard differences between 44.1 and 96 kHz in controlled tests, they have been listening over electrostatic headphones to specially generated test signals--various chirps and clicks, rather than real-world music, speech or even nature sounds (some of which are much more demanding than ordinary music)--and the people have generally had some training in how to listen critically to those test signals and hear differences. Some percussion instruments generate impulses that might be "diagnostic" for filter problems, though, plus we don't know whether playback systems might get better some day.
There is also a vague general belief that certain post-processing algorithms will generate less distortion if the sampling rate is higher. To me, if that is so, it sounds like a defect in the software--plus I've never seen anyone actually narrow it down to which algorithm or which software this is supposed to maybe happen with, i.e. I think we may be veering into urban-myth territory where that's concerned.
I still generally record at either 44.1 or 48 kHz when I record at all these days, depending on whether my recording will be synched up to video or not. But if I were a recording company (the two or three that are left nowadays) I suppose I would record at 96/24, because who knows--some day bandwidth may be so cheap and playback systems so good that it will matter a little to some people. I don't think that will happen during my lifetime, but that doesn't mean it can never happen.
--best regards
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If I read right, inputs are LINE, PIP 3/5V on the same jack. I know next to nothing about PIP. I gather when running PIP the input is mic level? If so, would I do any harm (physical or to the signal) running the mic level output from my Baby N-Box using this recorder?
I'm also interested to see how this works for .007 missions and getting past wands and walk-throughs. The metal case makes me a bit nervous, but I think the F3 uses metal too and that's being used in the field for that purpose I believe.
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I gather when running PIP the input is mic level?
Depends on the implementation, the details of which I don't think are known yet. However, on many small handheld recorders, the option of switching on/off PIP is separate from the choice of input sensitivity (mic/line). If that is the case with the PR-2, using line input sensitivity with PIP will be possible.
If so, would I do any harm (physical or to the signal) running the mic level output from my Baby N-Box using this recorder?
Probably not, but that depends on the circuit inside the N-Box. In all likelihood the N-box is designed to block DC (the PIP voltage from the recorder) applied to its output while allowing the alternating voltage signal of the audio to pass to the recorder.
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Just getting used to the F3 for my low security gigs with NBox Platinum but curious about the Deity until. For now my high >:D rig is babynbox dr-2d schoeps MK* but open to reports. Who says I can't have both.
You can have both (all three really), but they are very different animals. Not sure what a "low security" gig is (I assume that's still >:D but not 'open taping'? A gig you still have to sneak gear into?) Since the F3 has phantom power the N-Box platinum solution is quite a bit of extra gear over just using the off the shelf solutions like a CMC1L or CCM, which would eliminate the N-Box and extra cabling. Putting aside 32-bit for a moment, depending on your situation, using an F3, over a smaller cheaper common 2-channel recorder like an DR-2D ( or A10,M10, R07, etc) the advantage is not really clear. I'm not sure it outperforms the other recorders significantly enough on unbalanced line in to justify the extra gear. 32bit aside, F3 is certainly not the only one in its class.Centrance Mixerface comes to mind and is about the same size as the F3, The TASCAM DR40 is only slightly larger (and is slimmer and nearly entirely plastic), with the DR100 series stepping up from that in size a bit
Back to the subject of the thread, the Deity PR2 is really nothing like phantom recorders or the smaller line-in recorders. The former is larger and offers mic powering you may not need, the latter cant really power mics at all without additional equipment.
The PR2 looks to be the size of an A10 based on the pictures (size and weight not on product page but you can gauge by the size of the AA batteries). Timecode, 32-bit recording, and the ability to power mics at 5V in the form factor of one of the smallest 2-ch line in recorders is it advantage. First advantage is essentially useless to tapers, the second may be useful (especially given the level display on the pr2 appears to be as crippled as the TRX) but not really make-or-break, but the ability to properly power two mics at 5V in that form factor is a game changer. Nothing exists like that now. you need to step up to $1000+ solution to find that from Zaxcom, Lectrosonics, Sonosax, etc.... and most are larger and metal. the tentacle track-e comes close, but is only mono and youd need two devices at $600 and then have to sync them in post. Real 5V PIP power opens up proper powering at high-SPL of all lav mics, as well as the higher end DPA series, so more of a breakthrough for DPA guys. for Schoeps guys it seems to be the equivalent of just another A10 with 32-bit float replacing decent level meters.
In reality, implementation matters more than feature set. id rather have a reliable easy-to-use recorder with a battery box than an all-in-one that doesn't bring home the goods every time. So as you said.. i will wait for reports!
Well, would rather use the NBox Platinum with the actives I have than buying the CCM gear. Low security is stealth without having to navigate walkthrough or weapons scanners.
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Seems this will be delayed…
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Sounds like the price just went up as well
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(https://i.imgur.com/6rESN73.png)
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^LOL
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Too bad, definitely curious to see this one. The increase in price won't make a difference, the issue is sound and functionality are improved.
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Looks like they’ve now differentiated a US and worldwide model. Due to licensing patent tech from Zaxcom?
https://zaxcom.com/company/patents
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Nice /s gotta love Zaxcom
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I know Sonosax has had to release two versions of some of their recent recorders (maybe also the M2D2?) because of the Zaxcom patent. It's interesting that Stagetec, Sound Devices, Zoom, and Tascam do not. I wonder if the method by which Zaxcom gets 135 dB dynamic range from their preamps is not by using multiple auto-ranging DACs, or at least not implemented the same way as those other ones.
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^ Sound Devices, at least, has their own patent in this arena. If I recall, it is on the algorithm that they use to combine the streams from the multiple ADCs. Don't quote me on that, though (it has been a while since I read it)...
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I see deity have set up a page for a US version, https://deitymic.com/products/pr-2-us/
They write “US sold PR-2 units do not feature the mic pass thru option”, does anyone know what this means?
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From the non-us version of the page:
MIC PASS THRU
The PR-2 features a toggle switch on the top that allow you to pass the microphone PiP from the output thru to the input. This means you can daisy chain the PR-2 with a transmitter so you can add recording functionality to any transmitter you might own. This also means even if the PR-2 is turned off, the transmitters PiP is still passing thru to power your lavalier.
So my assumption is it's related to the patent discussion above, and you can't record and transmit at the same time. Probably also means you can't monitor live via the headphones, but that's less clear.
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I see deity have set up a page for a US version, https://deitymic.com/products/pr-2-us/
They write “US sold PR-2 units do not feature the mic pass thru option”, does anyone know what this means?
From the ex-US product page:
“The PR-2 features a toggle switch on the top that allow you to pass the microphone PiP from the output thru to the input. This means you can daisy chain the PR-2 with a transmitter so you can add recording functionality to any transmitter you might own. This also means even if the PR-2 is turned off, the transmitters PiP is still passing thru to power your lavalier.”
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At AES NY yesterday. The Deity booth didn't have much to show. No PR-2, not even a demo. No word on the price. May begin production in six weeks.
Meh.
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this year? :clapping:
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Sorry I haven't read the whole thread, how big is this? I can't find dimensions on the website.
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I've been keeping my eye on this recorder. I noticed that Gotham Sound mentions a Jan 2024 release date (link below). Not sure if this is firm or reliable.
https://www.gothamsound.com/product/pr-2-pocket-recorder
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i hear its at least as firm as july, august, september, october....
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I've been keeping my eye on this recorder. I noticed that Gotham Sound mentions a Jan 2024 release date (link below). Not sure if this is firm or reliable.
https://www.gothamsound.com/product/pr-2-pocket-recorder
I talked to a Gotham sales rep who I 've done some business with in the past back in September and he said they were firm on October but to anticipate delays - all manufacturers are experiencing delays due to supply chain challenges and then they went and changed the feature set meaning they might have had to redesign the hardware.
I've been watching this but I'm not holding my breath.
I went ahead and replaced the piggy back battery hack solution for my Korg MR1 so I'd have a small line in only recorder.
Honestly I may just keep doing that - it sounds really good and the battery solution is only $20 or so every few years.
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Well we are into January and still no pricing or delivery date
Why do companies do this?
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Gotham's the one who said it was a January release date, Deity has just said "coming soon". They're starting to deliver the Theos system, so hopefully the PR-2 isn't too far behind.
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They announced a change in the feature set - so likely a redesign - a few months back. Not surprised it's taking longer than expected. Reminds me of the Zoom F6. I waited so long for one of those I gave up and got something else then bought a used one for half price on Reverb six months after they were released.
I'm real interested in the Deity. I need a new small 2 channel recorder for lopro stuff. The Korg MR1 I've been using is still working and can do quite a long show after modifying the batteries but I'm not sure how much longer that thing will keep spinning the hard drive. Not having mics built in to it is very appealing for me.
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Same here. Would rather wait until they get it right given the price point.
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From ProAV in the U.K:
Thanks for your email.
I asked Deity and they’ve responded it is not available just yet, I will ask our web team to change the lead time to reflect this. Apologies for the inaccurate lead time. Hopefully it begins shipping soon, they did not provide me a date.
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https://www.gothamsound.com/product/pr-2-pocket-recorder
Gotham's website now says:
Model:
PR-2
Release date: April 2024
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https://www.gothamsound.com/product/pr-2-pocket-recorder
Gotham's website now says:
Model:
PR-2
Release date: April 2024
Follows the pattern they've been guessing. When date arrives, add 3 and guess again.
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I called Gotham earlier this week and learned that "Call for Price" is just an artifact of their website which defaults to that button when there is no price listed. I signed up for the notification on the Deity page for the PR-2 when it was announced in the middle of last April, and have heard NOTHING from them in nine months. Even if they decided to do a redesign this summer, I would expect the new parts to be priced by now, so $TBD even now is odd. When this was announced, it looked timely and interesting. Now, I'll likely wait for something better (96 kHz, 4 channel) unless the PR-2 sound is a clear improvement on the MMA-A.
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https://www.gothamsound.com/product/pr-2-pocket-recorder
Gotham's website now says:
Model:
PR-2
Release date: April 2024
Follows the pattern they've been guessing. When date arrives, add 3 and guess again.
Orders from certain “smaller mic builders that share a name where some people get married” ship faster than this thing. Zoom F3 it’ll be methinks.
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https://www.gothamsound.com/product/pr-2-pocket-recorder
Gotham's website now says:
Model:
PR-2
Release date: April 2024
Follows the pattern they've been guessing. When date arrives, add 3 and guess again.
Orders from certain “smaller mic builders that share a name where some people get married” ship faster than this thing. Zoom F3 it’ll be methinks.
Comparing Diety to that guy is about as low a blow as you can go. But warranted in this situation.
Face it folks; this thing is either vaporware or endlessly tied up in patent lawsuits.
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I think the PR2 will show up eventually, but the market for the theos is booming, and they had to redesign both because of the zaxcom patents.
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https://www.gothamsound.com/product/pr-2-pocket-recorder
Gotham's website now says:
Model:
PR-2
Release date: April 2024
Follows the pattern they've been guessing. When date arrives, add 3 and guess again.
Orders from certain “smaller mic builders that share a name where some people get married” ship faster than this thing. Zoom F3 it’ll be methinks.
Comparing Diety to that guy is about as low a blow as you can go. But warranted in this situation.
Face it folks; this thing is either vaporware or endlessly tied up in patent lawsuits.
To be fair, Church Audio products have actually been delivered and are in widespread use, unlike this vaporware from Deity. Their failure to communicate honestly is particularly galling, given that they are something more than a one-man operation.
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To be fair, Church Audio products have actually been delivered and are in widespread use, unlike this vaporware from Deity. Their failure to communicate honestly is particularly galling, given that they are something more than a one-man operation.
Key difference being Deity hasn't taken anyone's money, they've just announced they're working on the product. Gotham is the one giving bad dates. Deity just says "Coming Soon"
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Still bet they beat Boeing with the 737 Max 10. Need to make sure the SD card doesn't blow out. I will wait patiently.
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They're taking a while, but the Theos Transmitter is out, and the PR-2 basically looks like a pared down version that can't actually transmit? The Theos is a bit bigger, but presumably uses the same preamps/32-bit float system (and also outputs 3V or 5V), and I think it can just record instead of transmitting. I'd rather wait and spend less money, but I'm a bit surprised no one's put the Theos through its paces in a more rigorous way as a recorder.
https://deitymic.com/products/theos-digital-wireless/
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Has anyone used Deity products before? I've been taping off and on for more than 20 years and have never heard of them.
That doesn't necessarily mean anything, just curious.
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They started out making mics for "Content Creators", and expanded to wireless and recorders. Not focussed on tapers at all.
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They started out making mics for "Content Creators", and expanded to wireless and recorders. Not focussed on tapers at all.
No one focuses on "tapers" outside of a few small operations popular with people like us. As for major companies prioritizing location music recording, there are only a few left, even at the professional level.
Sound Devices, Zaxcom, Sonosax, Zoom,Tascam, Sony, and Roland seem to be it nowadays.
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Sound Devices, Zaxcom, Sonosax, Zoom,Tascam, Sony, and Roland seem to be it nowadays.
Although not music related, I immediately thought of Aaton. Just went to their website and saw this:
We are sad to announce that the Aaton-Digital adventure ended on February 15, 2024 by a court judgment.
Three years of covid-19 followed by a long strike by cinema professionals have profoundly affected our sales. Faced with no prospects, the company was liquidated.
We thank all our users and partners for their commitment and passion during the ten years we have gone through together.
edit- some discussion here: https://old.reddit.com/r/LocationSound/comments/1awq2f3/aaton_closed/
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Sound Devices, Zaxcom, Sonosax, Zoom,Tascam, Sony, and Roland seem to be it nowadays.
Although not music related, I immediately thought of Aaton. Just went to their website and saw this:
We are sad to announce that the Aaton-Digital adventure ended on February 15, 2024 by a court judgment.
Three years of covid-19 followed by a long strike by cinema professionals have profoundly affected our sales. Faced with no prospects, the company was liquidated.
We thank all our users and partners for their commitment and passion during the ten years we have gone through together.
edit- some discussion here: https://old.reddit.com/r/LocationSound/comments/1awq2f3/aaton_closed/
Yes I saw that posted on GS Remote the other day. The people who work with the Cantar machines swear by them.
Aeta stopped making the 4minx recorder awhile back, now only making interview / broadcast equipment. All Nagra recorders appear to be discontinued as well.
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They're taking a while, but the Theos Transmitter is out, and the PR-2 basically looks like a pared down version that can't actually transmit? The Theos is a bit bigger, but presumably uses the same preamps/32-bit float system (and also outputs 3V or 5V), and I think it can just record instead of transmitting. I'd rather wait and spend less money, but I'm a bit surprised no one's put the Theos through its paces in a more rigorous way as a recorder.
https://deitymic.com/products/theos-digital-wireless/
From my reading the transmitter only has a single input so you’d be recording in mono.
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From my reading the transmitter only has a single input so you’d be recording in mono.
There is a slew of devices like that - I guess you get what you pay for in terms of features and quality - I have an inexpensive Ulanzi system but have not really tested it to see what happens if I make a stereo recording using its two transmitters with or without connected external mics. But it would be a somewhat clumsy way of doing things. I also have a single DJI Mic 2, but being single, although it does 32 bit float, it's only mono.
Indeed, recording "just" audio these days seems to have become a bit of a olde worlde activity. These days, if I was asked to record an acoustic concert, and I offered the choice of audio or video with good-enough audio, I think the latter option would be popular. I wish I had the chance to dangle a remote-controlled camera with a four channel audio system plus external mic above an orchestra - which would cost somewhat less than some of the audio-only equipment that is rusting in my cupboard. The times they are a-changing...
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^The skill of focused visual imagination may be depreciating. At least the general appreciation of it seems to have. Excellent audio will conjure detailed visual imagery in the absence of visual stimuli, but the same doesn't occur in reverse. Hard to imagine a video-only corollary to audio-only radio. Silent era film I suppose, but even then my great-grandfather was playing fiddle with his sister on piano in the small home town film-theater. They reportedly knew just two or three instrumental themes and cycled them over and over again depending on what was going on in the scene, same music every film. I get that people tend to go for to "easy and obvious" over "quality" most of the time, but the thing that bothers me is how most folks gravitate to mediocre video+audio over really excellent audio on its own without the crappy video. I'm the opposite and tend to get quickly bored with mediocre video, whereas great audio perks me right up.
Decades ago it was easy, but convincing others of this now seems a near impossible sell. Still, a few years ago I successfully did so with a guy sitting next to me on a plane. He was watching a rather meh cell phone video of a concert that I happened to have made a good audio recording of and unusually, had on my phone. We started talking about the concert, about music, about taping, about the video, about audio recording quality, what matters, imagination, how it all fits together. He then skipped around watching segments of the first set of the concert, after which I suggested he listen to my audio-only recording of the second set through his same headphones. At first he was hesitant, but I insisted I would not be offended if he got bored at any point and wanted to bail back to the video. It was so fun to watch him settle into it, turning his head as if to look around at the concert environment he suddenly found himself immersed in, flashing a big thumbs up and a smile a the conclusion of the first tune. He ended up listening to the entire second set straight through, seemingly loving it. He couldn't say enough about it afterward as we deplaned and parted ways. That felt really good, but I've no doubt he went right back to shitty videos the following flight.
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This thread is sadly relevant to the above points:
https://gearspace.com/board/remote-possibilities-in-recording-amp-production/1424154-thinking-about-getting-out-remote-recording-altogether.html
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I do care about audio quality, but I've noticed that I perceive my own audio recordings to sound better than they really are when they're synced up with video. Also, when I rip audio from an official video stream to listen to it later, it often sounds worse than I remembered.
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I do care about audio quality, but I've noticed that I perceive my own audio recordings to sound better than they really are when they're synced up with video. Also, when I rip audio from an official video stream to listen to it later, it often sounds worse than I remembered.
In both of those cases, the audio has gone through lossy compression. Depending on what video editors you use, the default settings are not optimized for high quality audio.
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This thread is sadly relevant to the above points:
https://gearspace.com/board/remote-possibilities-in-recording-amp-production/1424154-thinking-about-getting-out-remote-recording-altogether.html
Ugh. 25 years ago I decided to not pursue recording music as a profession. That was difficult as it was where my heart was. Yet other avenues ended up being rewarding in different ways, and I still love recording music on my own terms today. Many roads, journeys, and destinations.
I do care about audio quality, but I've noticed that I perceive my own audio recordings to sound better than they really are when they're synced up with video. Also, when I rip audio from an official video stream to listen to it later, it often sounds worse than I remembered.
I notice the same, or something similar quite frequently. Its fascinating. There is definitely deeper perception stuff going on with this, which is more fundamental than lossy compression degradation. The second perceptual aspect presents a constant distraction from the focusing of critical perception on the other, acts as a crutch, and sets up a sort of perceptual bait and switch in our brains.
Back in the late 80's I spent hours soldering wiring/soldering switching units on site for a home/car stereo retailer. While doing so I'd sometimes tune an FM radio to the bottom end of the FM dial and pickup the audio from channel 6 TV. Fascinating to listen to some of the advertisements and syndicated TV shows sans video. Even with my concentration strongly focused on another task, my awareness of auditory details, sound editing, and all kinds of interesting auditory aspects was so interestingly heightened in comparison to listening while watching TV normally with the video content present. It was fun. Some ads didn't work at all. Some shows were terrible, others amazingly entertaining. Reruns of original Star Trek worked great and were almost like a radio drama, with much better radio-suited Foley than Gilligan's Island.
[/OT discussion from Diety PR-2]
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finally an update from andrew on ship date and price
https://www.youtube.com/watch?v=Nd7aGhDmNOM
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you got me
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:yack: :clapping:
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Well played, sir. Well played. :D
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In both of those cases, the audio has gone through lossy compression. Depending on what video editors you use, the default settings are not optimized for high quality audio.
I'll throw in a tiny mention of the DJI Pocket 3 video recorder, which can be set to record in wave format rather than embedding lossy AAC in the video file - but the source has to be its built in mics - although the stereo image from the 3 capsules is actually pretty good.
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Something ‘big’ is being announced on Sunday apparently.
https://deitymic.com/social/only-five-more-days-until-our-big-revealfree-nab-entry-using-our-code-ns7224deity-deitymics-deitymicrophones-bf1-dqc2-theos-theosduhf-uhf-duhf-spd1-nab-lasvegas/
(Probably nothing to do with the PR-2 though.
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it’s a transmitter for handheld and mini xlr microphones
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https://www.youtube.com/watch?v=Vs_lOPIZc-U
It's going to start shipping next month, BUT they've changed the specs: it will only support 32-bit float in MONO. Stereo is 24 bit only. If the pre-amps are clean, I can still see it being a very good recorder (small, 5V PiP), but the lack of 32-bit float is really disappointing after all this time waiting.
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https://www.youtube.com/watch?v=Vs_lOPIZc-U
It's going to start shipping next month, BUT they've changed the specs: it will only support 32-bit float in MONO. Stereo is 24 bit only. If the pre-amps are clean, I can still see it being a very good recorder (small, 5V PiP), but the lack of 32-bit float is really disappointing after all this time waiting.
Wha....?
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https://www.youtube.com/watch?v=Vs_lOPIZc-U
It's going to start shipping next month, BUT they've changed the specs: it will only support 32-bit float in MONO. Stereo is 24 bit only. If the pre-amps are clean, I can still see it being a very good recorder (small, 5V PiP), but the lack of 32-bit float is really disappointing after all this time waiting.
Wha....?
My exact reaction. They've updated the specs here: https://deitymic.com/products/pr-2/
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Also battery life: “Stereo: 15hrs (24-bit) with 16GB Card”
This is such a disappointment. I’ve been waiting for this to decide on buying and selling some equipment, and looking forward to a more organised inventory.
My Roland R-07 is better than this, it has the same battery life whilst recording a safety stereo track, which is the next best thing to 32-bit stereo recording. Simultaneous 24-bit stereo recordings, with one a few dB lower, is far more useful than a single mono 32-bit recording.
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I have 3 M-10's to wear out first. :lol:
If the PR-2 proves not to be vaporware I'll get one, I suppose...
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5V PIP, 24bit, 15hrs on 2 AAs, small device without attached mics. Still checks a lot of boxes.
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Well, it is made with filmmakers in mind who rarely need/use stereo, so not surprising but disappointing never-the-less from my perspective.
I should be hard pressed to buy a stereo non-32bit recorder at this point in time, but if everything else turns out to be perfect, it could be an interesting unit.
Deity products however usually needs a few firmware update iterations before they perform their best, so I don’t recommend being an early adopter - unless that’s your thing.
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https://www.youtube.com/watch?v=Vs_lOPIZc-U
It's going to start shipping next month, BUT they've changed the specs: it will only support 32-bit float in MONO. Stereo is 24 bit only. If the pre-amps are clean, I can still see it being a very good recorder (small, 5V PiP), but the lack of 32-bit float is really disappointing after all this time waiting.
32-bit is probably the least appealing feature for me
that interview is... something
its almost as if they dream up all these hypothetical products with feature sets, print cases for them, and dont even design working prototypes until a year later
extra bits wont buy you anything if the preamps are junk... "TDB", as they say.
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extra bits wont buy you anything if the preamps are junk... "TDB", as they say.
TDB... lol. Fitting, no?
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extra bits wont buy you anything if the preamps are junk... "TDB", as they say.
TDB... lol. Fitting, no?
Theyre kinda killin it these days!
maybe there is hope for Deity to save face
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also just because its 24-bit doesnt mean it cant be well implemented. take for example the dpa d:vice which is arguably one of the most adequate implementations of a basic pre and AD for our purposes. it uses the AKM5552VN which has 115dB of dynamic range
https://www.akm.com/content/dam/documents/products/audio/audio-adc/ak5552vn/ak5552vn-en-datasheet.pdf
a 5572 which still has relatively low power consumption gains 6 dB on this
https://www.akm.com/content/dam/documents/products/audio/audio-adc/ak5572en/ak5572en-en-datasheet.pdf
a 5578 with the channels summed can gain 6dB more, albeit being probably at the practical limits for power consumption
https://www.akm.com/content/dam/documents/products/audio/audio-adc/ak5578en/ak5578en-en-datasheet.pdf
good IC pres are cheap and common but it really depends on how much board space they can devote to a good analog gain stage. cleaner ADCs reduce the need for analog gain stage to an extent
all of this hits up against the wall of EIN of about 131dB. which easily fits into a 144dB 24bit container
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Guys...
I dont know how it is possible, but i get PR2 in next several days :o 8) :yahoo:
I ask seller where they get it - they told that their supplier in china set them ::) :hmmm:
It cost approx 280$ for me.
Do you know what's the funniest thing about all these? The fact that I live in Russia ::)
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it's priced at Gotham, with a June delivery date:
https://www.gothamsound.com/product/pr-2-pocket-recorder
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Like a fool, I got in on the pre-order queue.
Hurray for being a beta tester! :tomato: :banging head: :lol:
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Guys...
I dont know how it is possible, but i get PR2 in next several days :o 8) :yahoo:
I ask seller where they get it - they told that their supplier in china set them ::) :hmmm:
It cost approx 280$ for me.
Do you know what's the funniest thing about all these? The fact that I live in Russia ::)
congrats on being “the guy”
Don’t let us down with your review!
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Anyone know of a Europe/Canada/Japan dealer with the non-US version?
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
My first test will probably be 4015gs
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
Yup. That might be what I end up doing down the road if this thing delivers for when I need to do ultra stealth.
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16 hours of 24/48 is pretty compelling. set up, hit record and then turn it off when you get home
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
If you are recording two channels, it's a definite step backwards from the MMA-A -> iPhone (with Metarecorder) setup that's been around for six years. That will do 24/96 and, with my old iPhone, run over 10 hours. Unless the preamp or A/D clearly outclass the MMA-A, that would get me interested again. Waiting for a convincing review to say that.
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
If you are recording two channels, it's a definite step backwards from the MMA-A -> iPhone (with Metarecorder) setup that's been around for six years. That will do 24/96 and, with my old iPhone, run over 10 hours. Unless the preamp or A/D clearly outclass the MMA-A, that would get me interested again. Waiting for a convincing review to say that.
The sample rate limitation to 48 kHz is a bit odd, especially given that it can record in 24-bit fixed and 32-bit float formats. I wonder what chips they are using that can't do those things at 96 kHz.
Comparing specs, PR-2 EIN and dynamic range are 9 dB better than the MMA-A, but strangely, THD is worse by about the same amount. I will be interested to see some proper third-party measurements of this thing.
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That's a very attractive price point. And if it really does supply 5V, this tiny deck plus a set of DPA lavs would make a killer two-channel rig.
If you are recording two channels, it's a definite step backwards from the MMA-A -> iPhone (with Metarecorder) setup that's been around for six years. That will do 24/96 and, with my old iPhone, run over 10 hours. Unless the preamp or A/D clearly outclass the MMA-A, that would get me interested again. Waiting for a convincing review to say that.
The sample rate limitation to 48 kHz is a bit odd, especially given that it can record in 24-bit fixed and 32-bit float formats. I wonder what chips they are using that can't do those things at 96 kHz.
Comparing specs, PR-2 EIN and dynamic range are 9 dB better than the MMA-A, but strangely, THD is worse by about the same amount. I will be interested to see some proper third-party measurements of this thing.
100%
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It’s unboxing time!
At fist power on language was set to Chinese 🙈
It take some time (with help of google translate) to switch on BT and connect to Sidus app to change language)
I don’t know when i can test it live at some gig 🤷♂️
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It’s really tiny comparing to Zoom F1
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Thanks for posting these! Tiny for sure...
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looks nice!
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Can you confirm the PIP voltage?
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Can you confirm the PIP voltage?
DPA 4061 works)
Gain set to +24db and I spoke quite quietly
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Can you confirm the PIP voltage?
DPA 4061 works)
Gain set to +24db and I spoke quite quietly
Do you have a multimeter that reads DC voltage?
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Do you have a multimeter that reads DC voltage?
Nope (
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Waiting on reviews. Ready to buy one.
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given the limited track record of this company's products in our market vs the usual manufacturers, tbh i dont really get the people chomping at the bit for this, now that 32-bit is off the table, esp those who don't have a use for 5V pip. already numerous options for that including the trusty a10. take away 5v and were left with a device that we are hoping is, at best, equal to the A10 in its reliability and specifications, as A10 is quite a refined device that was a generation above the M10 and voice recorders it combined into a cost-effective, compact, high-resolution recording device with substantially cleaner input than its predecessors. last thing we need is another GIGO toy but time will tell.
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given the limited track record of this company's products in our market vs the usual manufacturers, tbh i dont really get the people chomping at the bit for this, now that 32-bit is off the table, esp those who don't have a use for 5V pip. already numerous options for that including the trusty a10. take away 5v and were left with a device that we are hoping is, at best, equal to the A10 in its reliability and specifications, as A10 is quite a refined device that was a generation above the M10 and voice recorders it combined into a cost-effective, compact, high-resolution recording device with substantially cleaner input than its predecessors. last thing we need is another GIGO toy but time will tell.
The fact that it doesn't have any mics on it will appeal to many people who want to just put it in their bag and have no questions asked.
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Yes thats certainly an upside. Franken-hacking an A10 to remove the mics isn't for everyone. I'm just concerned with the feature set changing at the 11th hour that this may not have an well designed analog front end. They wouldn't be changing an ADC chip this late in design (and most have been 32 bit-capable for years), so i fear with the limited real estate inside of there (massive volume taken up by batteries and a bunch of other stuff in there like timecode/wireless) that it isn't focused on clean 2-channel audio, more that it tries to be a jack-of-all-trades-master-of-none.
Modern 2ch recorders like the A10 and the DR100 utilize combining of channels on multi-channel ADCs to improve the S/N by 3 or more dB. The late-game design decisions/compromises make me believe they are at the limit of their hardware and it may be an FPGA-based design which can vary from exceptional to horrendous.
Time will tell, there is no substitute for testing,. teardown, and some real-world experience. we only had one report on the predecessor to this device and it did not meet the tapers expectations, hence why i said their track record is kinda iffy in our market.
The fact that they claim dynamic range in excess of 120dB but S/N at 90dB is also concerning. They dont go into measurement spec but 90dB SINAD would place it near the bottom of analog inputs in the modern day
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given the limited track record of this company's products in our market vs the usual manufacturers, tbh i dont really get the people chomping at the bit for this, now that 32-bit is off the table, esp those who don't have a use for 5V pip. already numerous options for that including the trusty a10. take away 5v and were left with a device that we are hoping is, at best, equal to the A10 in its reliability and specifications, as A10 is quite a refined device that was a generation above the M10 and voice recorders it combined into a cost-effective, compact, high-resolution recording device with substantially cleaner input than its predecessors. last thing we need is another GIGO toy but time will tell.
It would be a boring hobby/passion, if we all liked the same exact stuff. I have a lot of small decks and for me the A10 is one of my least favorites. That does not mean it is not a good deck, but I have lots of reasons I do not care for it. As small 24Bit recorders go, I far value the Roland R07 and R05 as well as the Marantz PMD 620 MKI and MKII. I still like the Tascam DR 2D as well, so I can see why lots would be eager to see what this deck can do. We really do not have much to go on yet, and it's not the specs, it's the performance and how people use their decks that often help decide what decks we like or do not like.
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Yes thats certainly an upside. Franken-hacking an A10 to remove the mics isn't for everyone. I'm just concerned with the feature set changing at the 11th hour that this may not have an well designed analog front end. They wouldn't be changing an ADC chip this late in design (and most have been 32 bit-capable for years), so i fear with the limited real estate inside of there (massive volume taken up by batteries and a bunch of other stuff in there like timecode/wireless) that it isn't focused on clean 2-channel audio, more that it tries to be a jack-of-all-trades-master-of-none.
Modern 2ch recorders like the A10 and the DR100 utilize combining of channels on multi-channel ADCs to improve the S/N by 3 or more dB. The late-game design decisions/compromises make me believe they are at the limit of their hardware and it may be an FPGA-based design which can vary from exceptional to horrendous.
Time will tell, there is no substitute for testing,. teardown, and some real-world experience. we only had one report on the predecessor to this device and it did not meet the tapers expectations, hence why i said their track record is kinda iffy in our market.
The fact that they claim dynamic range in excess of 120dB but S/N at 90dB is also concerning. They dont go into measurement spec but 90dB SINAD would place it near the bottom of analog inputs in the modern day
It's all speculation/what-if until somebody as a proper teardown and can vouch for what it truly is/isn't. I'm not holding my breath on this so I went with an F3 which made more sense to me despite it being "larger."
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no matter what the other differences are, the f3 isn't "larger", it's significantly larger. And has significantly worse battery life.
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no matter what the other differences are, the f3 isn't "larger", it's significantly larger. And has significantly worse battery life.
Depends what youre into. I would probably never use F3 in a situation where I didnt need either phantom or the ability to take a hot sbd feed. I see tapers using F3 with unbalanced stealth rigs as if its some improvement and don't get it. if you can get all your gear in and run it yet dont know your gear well enough to lean on (what is essentially a crutch) of 32 bit to capture your data, you're probably missing the mark. The only time i ever use 32bit float on the devices i have is if its completely unattended which is not the case with gear on your body. adjusting levels is as fundamental as ensuring you remembered to press record, methinks.
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I'm interested in this for a specific application, but will be waiting for test reports from other users to determine if it will actually be capable of doing what I need, which is recording 4 to 6 channels using 2 or 3 of these units in combination, with centralized remote control and sync.
Other than sufficient powering and SINAD to accommodate a pair of DPA CORE 4060 [the baseline go/no-go performance metric for me], the critical functionality for my use will be if two or three of them can be sufficiently sync'd and controlled while in pocket, using the software below or something equivalent to it installed on my phone..
seems you can control the units with this software
https://www.sidus.link/sidusAudio/software
The goal is replacing: 4 X 4060 > 4 channel CA-UGLY2 > DR2d. I need at least 4, or better 6 channels (modular addition good), operable from a single control. I would much prefer all 4-6 channels to a single recorder, but there is nothing small enough capable of that. Given the elimination of an external preamp or bat box, I can deal with 2 or 3 of these units in pocket, which shoud be no larger than CA-UGLY2 > DR2d, and probably somewhat better form wise.
Control and sync will be the key.
If you come across reviews which include sync control of multiple units, please post a link. Thanks.
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FYI to further the conversation, I've also been thinking of picking up an F3 for soundboard / 2nd small 2ch phantom rig use.
Different use case, different feature set, different horses for courses.
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no matter what the other differences are, the f3 isn't "larger", it's significantly larger. And has significantly worse battery life.
Depends what youre into. I would probably never use F3 in a situation where I didnt need either phantom or the ability to take a hot sbd feed. I see tapers using F3 with unbalanced stealth rigs as if its some improvement and don't get it. if you can get all your gear in and run it yet dont know your gear well enough to lean on (what is essentially a crutch) of 32 bit to capture your data, you're probably missing the mark. The only time i ever use 32bit float on the devices i have is if its completely unattended which is not the case with gear on your body. adjusting levels is as fundamental as ensuring you remembered to press record, methinks.
The 32-bit float part here is just not true? I've seen it repeated here several times that 24-bit has more than high enough dynamic range not to be a problem in itself, and this IS true - but only if you have multiple ADCs to go along with it. The truth of the matter is that depending on the type of music you like, taping shows is too unpredictable for you to nail your gain settings for every part of the show. I can record a quiet song with no apparent preamp noise using a quality preamp's higher gain setting (EIN gets higher the higher gain you use); I can also record a louder song by dialling the gain down (EIN decreases, but this doesn't really matter because the signal is strong compared to the preamp's noise floor). What I can't do is use the low gain setting for BOTH sections of music (the only option we have when we don't know when they're coming), and expect the less-than-ideal staging for the quiet parts to sound good. Of course this is only really relevant if you're using compression or amplifying certain songs in post, but some bands really do get that loud AND that quiet, and it's a terrible experience to listen back without narrowing the dynamic range.
The entire point of these 32-bit float devices is that they have multiple ADCs that attain good gain staging for ALL parts of the music. The issue was never the noise floor inherent to the 24 bit format (which exists, but is too quiet to matter). The issue is clipping when you're trying to optimise your analogue gain for the quietest sounds you're trying record. A Zoom F3 bypasses this issue altogether, which is why people are so fond of using it. It isn't simply a crutch. Are you seriously adjusting your analogue gain mid song all the time? Unless you know exactly what's coming, that's not practical, and the level differences you get by doing it by hand are very annoying to fix in post.
I say this as someone who doesn't have an F3. If it had a minijack input, or a longer battery life, I'd go for it in a heartbeat.
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no matter what the other differences are, the f3 isn't "larger", it's significantly larger. And has significantly worse battery life.
Lol. Christ, I had a WM-D3 back in the day and a TCD-D7 as well. The F3 doesn't phase me.
Battery life, even with P48 on, can be between 3.5h with IKEA Ladda batteries... If you use Energizer Lithium batteries, you can double that. Plenty for my needs.
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The 32-bit float part here is just not true?
I'm thinking you missed the part where they revised the specs that the unit no longer does 32-bit float when used for 2 channel recording? single channel 32-bit float recorders with 5V power for lavs have existed for awhile (Tentacle Track E) so the PR2 doesnt break any ground in that regard.
In regard to the rest of the gain-ranging discussion, id have to respectfully disagree with your take. Weve already seen some units like the zoom raise the noise floor when switching ADCs (something that it seems most people agree is inconsequential to our use). In the same manner, the "wide dynamic range" of the zoom series is really limited to +4 on the top end (and the typical modern input approaching -130dB EIN. When you switch it from mic to line in to allow for hotter signals it just throws a pad on and you lose ~20dB of dynamic range. I have yet to see a single measurement that shows a 32-bit float recording offering increased dynamic range over a 24-bit counterpart, as all the 32 bit equipment ive measured or seen tested (admittedly only zoom and SD) has a dynamic range that fits easily inside the 144dB 24bit container.
for audience recording purposes a 24-bit recording has limited use over a 16-bit recording as were usually working with 60-70 dB dynamic range at most between crowd noise/wind/hvac/etc
In this case i just dont see 32bit float being at all useful for a device with 90dB SINAD, let alone the (presumably) cleaner zoom and SD units (which have all been tested and shown to have marginal inputs at best to save to 32-bit). yeah it might save your recording if you blow your levels, but you know what else will... a 24 bit recording peaking at -12dB
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FYI to further the conversation, I've also been thinking of picking up an F3 for soundboard / 2nd small 2ch phantom rig use.
Different use case, different feature set, different horses for courses.
I've used my F3 to plug into a handful of boards and it's great to not worry about checking it while I video.
I have also noticed at several venues the USB Thumb drive mix, my main mix if available, differs from the XLR SBD mix.
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I can record a quiet song with no apparent preamp noise using a quality preamp's higher gain setting (EIN gets higher the higher gain you use); I can also record a louder song by dialling the gain down (EIN decreases, but this doesn't really matter because the signal is strong compared to the preamp's noise floor). What I can't do is use the low gain setting for BOTH sections of music (the only option we have when we don't know when they're coming), and expect the less-than-ideal staging for the quiet parts to sound good. Of course this is only really relevant if you're using compression or amplifying certain songs in post, but some bands really do get that loud AND that quiet, and it's a terrible experience to listen back without narrowing the dynamic range.
this is all simple to do in post, without any guessing. As you mentioned, you raise the noise floor when you raise the gain in the field. thats fixed to your mic self noise and room noise, all of which gets amplified whether you do it in real time or post.
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I can record a quiet song with no apparent preamp noise using a quality preamp's higher gain setting (EIN gets higher the higher gain you use); I can also record a louder song by dialling the gain down (EIN decreases, but this doesn't really matter because the signal is strong compared to the preamp's noise floor). What I can't do is use the low gain setting for BOTH sections of music (the only option we have when we don't know when they're coming), and expect the less-than-ideal staging for the quiet parts to sound good. Of course this is only really relevant if you're using compression or amplifying certain songs in post, but some bands really do get that loud AND that quiet, and it's a terrible experience to listen back without narrowing the dynamic range.
this is all simple to do in post, without any guessing. As you mentioned, you raise the noise floor when you raise the gain in the field. thats fixed to your mic self noise and room noise, all of which gets amplified whether you do it in real time or post.
That's the thing though, you are assuming the main source of noise is the mic self noise and room noise. But I can consistently hear the preamp self-noise before either of those, when we're talking about recorders like the Zoom H series, or the Roland R-05, or the Sony PCM-M10. The Zoom H1 and H1N, for example, automatically switch circuits when you increase your gain past +13dB or so - you can clearly hear a drop in hiss when you go from +12 dB to +13 dB, even though you're getting more gain. The same is true of the Roland's MicLOW and MicHI gain settings: their gain range overlaps, so if you set both to the same point in their overlapping range, MicHI will very clearly sound less noisy, and that becomes particularly apparent when you add even more gain digitally in post.
The thing about the Zoom F3 is that not only does it have cleaner pre-amps than the aforementioned recorders, but it also works in such a way that you don't have to pick between MicLOW and MicHI, it will just switch automatically as needed. Maybe our use cases are too different, I don't know. What I'm saying is particularly true when you're using mics that aren't too sensitive and you aren't using any external pre-amps, so you're always using Mic instead of Line In (I appreciate your point that if your signal is hot enough, the F3 just applies a pad, but my mics without an external pre-amp are basically never hot enough).
I agree this discussion is not particularly relevant to the Deity PR-2, except for the fact that supposedly its pre-preamp outperforms all the recorders I've mentioned except the Zoom F3 and the MixPre. At this point the advantage of the Deity over the worse recorders I've mentioned is the size, battery life, 5V PiP and MAYBE pre-amp quality, but that's it.
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That's the thing though, you are assuming the main source of noise is the mic self noise and room noise. But I can consistently hear the preamp self-noise before either of those, when we're talking about recorders like the Zoom H series, or the Roland R-05, or the Sony PCM-M10.
fair enough. i would personally never use any of those devices as preamps so i dont really have that experience
i cant hear noise in A10 in the cases ive used it as a pre with small electrets, but it might be old ears...
I agree this discussion is not particularly relevant to the Deity PR-2, except for the fact that supposedly its pre-preamp outperforms all the recorders I've mentioned except the Zoom F3 and the MixPre.
thats speculative at this point, we will see! i hope its great, but my expectations are tempered. we can hope its well above the zoom H series which is known for being nosiy.
the more i dig the 90 dB sinad isnt all that uncommon on mic ins and isnt far from sound devices own spec. not state-of the art but"good enough" for our purposes, perhaps we ahve a case of transparently honest measurement, which would be nice
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fair enough. i would personally never use any of those devices as preamps so i dont really have that experience
i cant hear noise in A10 in the cases ive used it as a pre with small electrets, but it might be old ears...
Yeah, I avoid using them as preamps when possible too! The allure of the PR-2 is the promise of 5V PiP and the hopes that the preamp will be good enough for you to ditch, say, a CA9200 preamp. If you still need to use an external preamp, 5V PiP is kind of pointless anyway.
thats speculative at this point, we will see! i hope its great, but my expectations are tempered. we can hope its well above the zoom H series which is known for being nosiy.
the more i dig the 90 dB sinad isnt all that uncommon on mic ins and isnt far from sound devices own spec. not state-of the art but"good enough" for our purposes, perhaps we ahve a case of transparently honest measurement, which would be nice
Yeah, we'll see! I hope it's good, but who knows at this point.
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The 32-bit float part here is just not true?
I test it on my PR2 and can confirm that 32bit works ONLY in mono mode. So when you switch to stereo you also switch to 24bit. No way to select 32bit and stereo :bawling:
I'll try to visit gig on 8th of june to test PR2 in the wild >:D
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I have to admit i don't understand the utility of "mic pass thru" (which is disabled on US models). If the unit provides PIP, why would you ever need to pass PIP from a transmitter thru the output to the input? Couldn't you power the mic from the PR2 and send a line signal to the transmitter, without violating any patents?
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Other than sufficient powering and SINAD to accommodate a pair of DPA CORE 4060 [the baseline go/no-go performance metric for me], the critical functionality for my use will be if two or three of them can be sufficiently sync'd and controlled while in pocket, using the software below or something equivalent to it installed on my phone..
metarecorder can do this but not elegantly, via its master/slave mode. they are all clocked independently though, same as the deity PR2s would presumably, as time code sync is not the same as word clock.
not elegant because you need a separate phone for each 2 channels. i think only the teenage engineering unit that hoserama runs fits your bill but thats a monster with a price to match
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I have to admit i don't understand the utility of "mic pass thru" (which is disabled on US models). If the unit provides PIP, why would you ever need to pass PIP from a transmitter thru the output to the input? Couldn't you power the mic from the PR2 and send a line signal to the transmitter, without violating any patents?
This is a function needed by the crowd that this recorder is primarily marketed to - which is NOT concert tapers.
From the Deity website -
"MIC PASS THRU
The PR-2 features a toggle switch on the top that allow you to pass the microphone PiP from the output thru to the input. This means you can daisy chain the PR-2 with a transmitter so you can add recording functionality to any transmitter you might own. This also means even if the PR-2 is turned off, the transmitters PiP is still passing thru to power your lavalier."
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Seems like that's not a useful feature anyway, if the pr2 will provide the PIP. Just send output back to the transmitter without sending PIP forward
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Other than sufficient powering and SINAD to accommodate a pair of DPA CORE 4060 [the baseline go/no-go performance metric for me], the critical functionality for my use will be if two or three of them can be sufficiently sync'd and controlled while in pocket, using the software below or something equivalent to it installed on my phone..
metarecorder can do this but not elegantly, via its master/slave mode. they are all clocked independently though, same as the deity PR2s would presumably, as time code sync is not the same as word clock.
not elegant because you need a separate phone for each 2 channels. i think only the teenage engineering unit that hoserama runs fits your bill but thats a monster with a price to match
Right. I've no interest in recording to 2 or 3 separate phones, which I could do using DPA d:vice. I was long hoping for some way of connecting 2 or 3 d:vice's to a single phone. Or a follow up d:vice version that featured a card slot able to recorded locally, but with collective control over several. The speculative strategy of recording using 2 or 3 PR-2's that are collectively controlled by a single phone is based on that. Nice to have the storage local, only remaining problem if that works is clock-sync.
Non-fully sync'd clocks is unfortunate, yet hopefully easily manageable, which is the key to this working as envisioned. Yes timecode is not wordclock. The difference between frame rate and sample rate is around 2 orders of magnitude! However, I've found modern clock chips, especially in gear which features timecode to free-run with a quite tight tolerance, particularly when its the same chip in identical recorders. I'm holding out hope the drift will be minimal, perhaps inconsequential over the course of one set of music.
^
Getting into the weeds on this a bit- In one potential arrangement I'd need 5 channels total, 3 of which need be closely phase-correlated. One way of checking / assisting alignment is using the pass-thu feature to duplicate one channel from the first recorder which is recording two mic inputs to one channel of the second recorder that's only recording one mic. The duplicate track can then be used to confirm sync, and if necessary assist with stretching in a very accurate way (invert phase and sum with the duplicate, alignment achieves deepest null). I've done that a couple times using two four-channel DR2d's to record 6 channels by routing the headphone out from the first recorder to one of the two stereo inputs on the second recorder, duplicating that pair between the two recorders. Technically this incurred a very slight but constant latency offset due to the DA/AD step between the first to second recorders. But that wasn't a problem, and if it were it would be easily measurable and would not change. Of course, I'm hoping that's unnecessary, but its a work around.
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[snip..] The truth of the matter is that depending on the type of music you like, taping shows is too unpredictable for you to nail your gain settings for every part of the show. I can record a quiet song with no apparent preamp noise using a quality preamp's higher gain setting (EIN gets higher the higher gain you use); I can also record a louder song by dialling the gain down (EIN decreases, but this doesn't really matter because the signal is strong compared to the preamp's noise floor). What I can't do is use the low gain setting for BOTH sections of music (the only option we have when we don't know when they're coming), and expect the less-than-ideal staging for the quiet parts to sound good. Of course this is only really relevant if you're using compression or amplifying certain songs in post, but some bands really do get that loud AND that quiet, and it's a terrible experience to listen back without narrowing the dynamic range. [..snip]
In practice, I don't find the first part to be a constraint in my experience, yet fully agree that some content really needs a reduction in dynamic range to be enjoyably listenable.
Using the 4 X DPA_4060 > 4ch_CA-UGLY2 > Dr2d rig I mentioned, I find I can successfully use just two different gain settings on the preamp to accommodate everything I record, and don't need to switch between them during any single performance [edit- It did require some experimentation to determine those two settings, but one determined the two are all I need]. The higher gain setting is used for anything with an exceptionally low acoustic noise floor such as some classical music and small acoustic jazz performances, yet has still proven capable of accommodating highly dynamic full orchestra performances at point-blank range. The lower gain setting I use for PA amplified concerts which almost always have a significantly higher ambient noise floor and do not suffer from the higher EIN, or when in close proximity to drum kits where the dynamic transients can be extreme making the additional headroom attractive. That latter case is the one that could potentially be problematic, but fortunately I've still not had EIN exceed the ambient noise floor in any of my real-world situations.
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Non-fully sync'd clocks is unfortunate, yet hopefully easily manageable, which is the key to this working as envisioned. Yes timecode is not wordclock. The difference between frame rate and sample rate is around 2 orders of magnitude! However, I've found modern clock chips, especially in gear which features timecode to free-run with a quite tight tolerance, particularly when its the same chip in identical recorders. I'm holding out hope the drift will be minimal, perhaps inconsequential over the course of one set of music.
I can confirm from experience that the modern clocks in the Zoom F Series appear to not drift apart; many times I've ran a F6 onstage and an F3 in the audience and every single time I've inserted those into multitrack software, it only requires syncing up the 2 independent sources at the beginning, usually the very first snare hit of the show. Then I scroll to the end of the show, and sure enough to my amazement, the final snare hit (or whatever) IS RIGHT ON THE MONEY. Even had a couple 4+ hour shows and same result. I couldn't believe it at first but I've done this 7-8 times now and every single one was drift-free.
I had one show where I ran an F6, F3, and F2 (mono) and all 3 independent sources didn't drift apart!
After hearing about Deity removing 32FP from stereo mode, I am going to try to acquire a 2nd F2 and run 2 of them for stereo >:D. Since the F2 is only 2.5v PiP, I plan on getting a cable like this:
https://www.amazon.com/CablesOnline-3-5mm-Stereo-Breakout-AM-603C/dp/B0759Z1QCP/ref=asc_df_B0759Z1QCP/?tag=hyprod-20&linkCode=df0&hvadid=693461287397&hvpos=&hvnetw=g&hvrand=9649726971229560676&hvpone=&hvptwo=&hvqmt=&hvdev=c&hvdvcmdl=&hvlocint=&hvlocphy=9028778&hvtargid=pla-360555372916&psc=1&mcid=c9f4eaa0e00c3021bde9cf841429c5ab&gad_source=1 (https://www.amazon.com/CablesOnline-3-5mm-Stereo-Breakout-AM-603C/dp/B0759Z1QCP/ref=asc_df_B0759Z1QCP/?tag=hyprod-20&linkCode=df0&hvadid=693461287397&hvpos=&hvnetw=g&hvrand=9649726971229560676&hvpone=&hvptwo=&hvqmt=&hvdev=c&hvdvcmdl=&hvlocint=&hvlocphy=9028778&hvtargid=pla-360555372916&psc=1&mcid=c9f4eaa0e00c3021bde9cf841429c5ab&gad_source=1)
and running that out of a batt box and then into the 2 mono recorders. I'll probably start the recorders at home or in the car, clap or dog-click to give me an initial peak to line-up, and then do it once more after the show to confirm any drift. Even if it does drift, the claps/dog-clicks will allow a very simple and easy stretch. Lining up claps directly in front of your mics in a quiet car are VERY easy to work with compared to snare hits or drum-less performances.
I've only >:D one show with my F2, a classical show, and I'll try to dig it up later tonight if anyone wants to hear the preamp. It's a Zoom F series, although it's sooo tiny (smaller than the PR-2?) I'm curious if it performs as well as an F3 etc.
The Zoom website says 15 hours on AAA alkalines, so I'm guessing 8-10 hours on rechargeables? I haven't tested it yet.
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Deity PR2 vs Zoom F2 BT )))
Zoom F2 a little bit smaller but mono only 😢
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After hearing about Deity removing 32FP from stereo mode, I am going to try to acquire a 2nd F2 and run 2 of them for stereo >:D.
If you do, let us know how well that works. Be aware that it can to be somewhat more of a challenge to split a single stereo pair across two recorders than it is to record two separate sources that will be mixed together to separate recorders. At least some stereo pairs where maintaining tight phase correlation between the two channels is more critical, such as a coincident pair. Should be less of an issue with a wide-spaced pair.
On the wide-pair front, that should make for a great wide-split stereo rig that uses a single mic + tiny recorder attached to the upper side balcony rail near the PA, with the same duplicated on the opposite side of the room/stage, with no wires between them.. an arrangement speculated about in some other recent threads (as well as some old ones). Although it would be arguably even better to use two stereo recorders to do the same, with the second mic of each pair facing away from the PA and out into the room, so as to provide control over the ratio of direct PA sound verses ambient/reverberant/room afterward.
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Be aware that it can to be somewhat more of a challenge to split a single stereo pair across two recorders than it is to record two separate sources that will be mixed together to separate recorders. At least some stereo pairs where maintaining tight phase correlation between the two channels is more critical, such as a coincident pair. Should be less of an issue with a wide-spaced pair.
Yea I wondered about the lack of precision when lining up independent sources in post. Does it affect critical time arrival differences? I'm sure it does. Probably easier to just wait for the first micro stereo 32fp recorder. We'll see.
Thanks for the size comp pics.
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If anyone in Europe/Canada finds this for sale and it's the 32 bit stereo version, please post and I'll order one
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If anyone in Europe/Canada finds this for sale and it's the 32 bit stereo version, please post and I'll order one
there is no 32-bit stereo version. only difference in US version is no PIP pass thru
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If anyone in Europe/Canada finds this for sale and it's the 32 bit stereo version, please post and I'll order one
there is no 32-bit stereo version. only difference in US version is no PIP pass thru
Oh, I thought it was said that the non-US versions did not change from original "specs". Thanks
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It's not entirely clear, at least to me. On the non-US version it says "Supports 24-bit and 32-bit float recording formats" at the top of the page and "32-bit Float Mono / 24-bit (Mono/Stereo)" under specs. The US version has the same line in the specs, but says "Supports 24-bit Stereo and 32-bit float Mono recording formats" at the top. So there is an error somewhere, but, barring any additional information, who knows where...
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from the page today for the non-us version:
32-bit Float Mono / 24-bit (Mono/Stereo)
https://deitymic.com/products/pr-2/
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Yeah, a little spec confusion going around...
Psinka - can you verify if your version can record in 32 bit stereo?
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Yeah, a little spec confusion going around...
Psinka - can you verify if your version can record in 32 bit stereo?
I already test these: when you select 32bit channels goes to mono and no way to select stereo with 32bits.
32 bits accessible for mono only. Stereo can be recorded with 24bit only 🤷♂️
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Yeah, a little spec confusion going around...
Psinka - can you verify if your version can record in 32 bit stereo?
I already test these: when you select 32bit channels goes to mono and no way to select stereo with 32bits.
32 bits accessible for mono only. Stereo can be recorded with 24bit only 🤷♂️
Thanks for confirming!
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from the page today for the non-us version:
32-bit Float Mono / 24-bit (Mono/Stereo)
https://deitymic.com/products/pr-2/
yeah i dont think that page has been updated since well before the spec change was announced last month. i cant think of a reason they would lock out 32-bit stereo recording in the US
note that the box that Psinka has also says "32-bit float recording: mono/stereo" right on the box
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Be aware that it can to be somewhat more of a challenge to split a single stereo pair across two recorders than it is to record two separate sources that will be mixed together to separate recorders. At least some stereo pairs where maintaining tight phase correlation between the two channels is more critical, such as a coincident pair. Should be less of an issue with a wide-spaced pair.
Yea I wondered about the lack of precision when lining up independent sources in post. Does it affect critical time arrival differences? I'm sure it does. [snip..]
Somewhat. I think the issue will be if the error in precision falls within our perceptual tolerance for error. When aligning the two channels of a coincident pair and you zoom in, the peaks of both channels will/should always align, whereas with a split pair it's easy to see the time offset of peaks arriving from sources that are off-center. Sometimes you just need to split the difference when aligning a split pair with SBD. Additionally, perfect alignment is somewhat less critical when aligning two separate stereo sources, than when aligning the two channels of a single stereo pair.
On that last point but even more OT, there is a technique proposed maybe 20 years ago for mixing in spot mics with a main mic pair, which might be applicable to mixing in SBD with AUD, since a SBD feed acts essentially as a "collective spot mics" stem. It seeks to better preserve the imaging and depth cues of the main/AUD pair by delaying the spot/SBD signal by a few different randomized milliseconds or factions of, and pans those slightly delayed signals around to random static locations so they act like early reflections. The main pair/AUD source then retains unmolested first arrival status preserving its image cues, with the pseudo early reflections provide clarity and "direct-soundiness". Never tried that, but its an interesting approach modeled on the behavior of real world direct/early/late arrival acoustics.
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Yeah, a little spec confusion going around...
Psinka - can you verify if your version can record in 32 bit stereo?
I already test these: when you select 32bit channels goes to mono and no way to select stereo with 32bits.
32 bits accessible for mono only. Stereo can be recorded with 24bit only 🤷♂️
Dealbreaker. Tascam DR-2D I use already does that.
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…and then some (4ch)
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The more I learn about this recorder, the more I'm confused by it. What kind of customer is this being marketed towards? It has too many limitations (sample rate, EIN) to be considered a professional-tier unit, IMO. A bait-and-switch on 32FP only being available in mono is not a pro move.
I think the only good option in a similar form factor is the Lectrosonics SPDR. You get their "split gain" when recording in 48 kHz. The biggest negative with the SPDR is only 4V PIP.
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I think the only good option in a similar form factor is the Lectrosonics SPDR. The biggest negative with the SPDR is it's $1400 price tag
The Deity is clearly being marketed towards audio for prosumer videographers.
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I think the only good option in a similar form factor is the Lectrosonics SPDR. The biggest negative with the SPDR is it's $1400 price tag
The Deity is clearly being marketed towards audio for prosumer videographers.
I guess they want some of the Zoom / Tascam market. The PR-2 seems to be a mix of the Zoom F1 and F2 recorders.
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Yeah, a little spec confusion going around...
Psinka - can you verify if your version can record in 32 bit stereo?
I already test these: when you select 32bit channels goes to mono and no way to select stereo with 32bits.
32 bits accessible for mono only. Stereo can be recorded with 24bit only 🤷♂️
I am still unclear. Because the unit does record in stereo, and because it does accept a stereo mic, what exactly does it mean when 32Bits is selected? Does it go to 2 mono channels? Does it go to one mono track only? If so it it pulling from the left mic or the right mic. I am just looking for some details to understand what switching to mono in 32Bit mode means exactly.
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It means mono has 24 and 32 bit options and stereo only has 24
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It means mono has 24 and 32 bit options and stereo only has 24
That in no way answers my questions.
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I am still unclear. Because the unit does record in stereo, and because it does accept a stereo mic, what exactly does it mean when 32Bits is selected? Does it go to 2 mono channels? Does it go to one mono track only? If so it it pulling from the left mic or the right mic. I am just looking for some details to understand what switching to mono in 32Bit mode means exactly.
I don't have it, but I assume it will just pull from the left mic (the tip connector), since that's the only connector mono mics also have.
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It means mono has 24 and 32 bit options and stereo only has 24
That in no way answers my questions.
I thought that was a very clear response. In 24-bit mode, it can record two channels. In 32-bit mode, it can only record one. As to why this is the case, that's a mystery.
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My speculation, yet seems clear enough to me. The auto-switching between multiple input gain / ADC stages that happens in the course of constructing the data for the 32 bit file requires two or more separate gain/ADC paths (3 in the case of SoundDevices). A stereo path provides two. They can use them as two separate channels with identical gain for 24bit stereo, or use both for a single mono channel by setting them to different gains and auto-switching between them to produce the data which gets written as a mono 32 bit file.
Unfortunately it has not been made clear by the manufacturers, but the auto-gain switching part and the 32-bit floating point file storage format are really two entirely separate things. In addition to the typical way of using a switching input path and writing a 32bit file from that, its entirely possible to use a single non-switching input stage /ADC and produce a 32bit floating point file. Or to use a switching input stage design and write a 24bit fixed file from that. The first is equivalent to using a traditional recording interface and writing a 32bit float file in the recording software. The second is how the current 32bit recorders work in 24bit mode.
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I’m hopeful that the hardware is there and later firmware updates will add the functionality. But if not, it still fills a need for me.
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It means mono has 24 and 32 bit options and stereo only has 24
That in no way answers my questions.
I thought that was a very clear response. In 24-bit mode, it can record two channels. In 32-bit mode, it can only record one. As to why this is the case, that's a mystery.
It may be clear but does not answer my question. My question was specific to someone who has the unit. The above answer is obvious and not what I was asking. Perhaps repeating what I am asking will help.
Because the unit does record in stereo, and because it does accept a stereo mic, what exactly does it mean when 32Bits is selected? Does it go to 2 mono channels? Does it go to one mono track only? If so it it pulling from the left mic or the right mic. I am just looking for some details to understand what switching to mono in 32Bit mode means exactly.
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They have not said their methodology for the 32 bit encoding, and typically a mono recording would be the tip and sleeve from the minijack, which would be left channel, one track track
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It means mono has 24 and 32 bit options and stereo only has 24
That in no way answers my questions.
I thought that was a very clear response. In 24-bit mode, it can record two channels. In 32-bit mode, it can only record one. As to why this is the case, that's a mystery.
It may be clear but does not answer my question. My question was specific to someone who has the unit. The above answer is obvious and not what I was asking. Perhaps repeating what I am asking will help.
Because the unit does record in stereo, and because it does accept a stereo mic, what exactly does it mean when 32Bits is selected? Does it go to 2 mono channels? Does it go to one mono track only? If so it it pulling from the left mic or the right mic. I am just looking for some details to understand what switching to mono in 32Bit mode means exactly.
My apologies - I was not understanding your question.
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My speculation, yet seems clear enough to me. The auto-switching between multiple input gain / ADC stages that happens in the course of constructing the data for the 32 bit file requires two or more separate gain/ADC paths (3 in the case of SoundDevices). A stereo path provides two. They can use them as two separate channels with identical gain for 24bit stereo, or use both for a single mono channel by setting them to different gains and auto-switching between them to produce the data which gets written as a mono 32 bit file.
That theory sounds plausible, if a bit strange implementation. AFAIK, all other modern field recorders with multiple ADCs have the same number of channels flowing through each of them regardless of the file format eventually being written.
The 48 kHz limitation with this device in either mode still has me scratching my head.
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^ I think the film standard is 48 kHz?
With respect to the 24-/32-bit issue, I think it requires quite a bit of processing. I suspect they either cheaped out or were unable to obtain their desired chip.
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My speculation, yet seems clear enough to me. The auto-switching between multiple input gain / ADC stages that happens in the course of constructing the data for the 32 bit file requires two or more separate gain/ADC paths (3 in the case of SoundDevices). A stereo path provides two. They can use them as two separate channels with identical gain for 24bit stereo, or use both for a single mono channel by setting them to different gains and auto-switching between them to produce the data which gets written as a mono 32 bit file.
That theory sounds plausible, if a bit strange implementation. AFAIK, all other modern field recorders with multiple ADCs have the same number of channels flowing through each of them regardless of the file format eventually being written.
The 48 kHz limitation with this device in either mode still has me scratching my head.
as I mentioned earlier in the thread all of the chip manufacturers have multichannel chips that can be summed as needed, but a 4 and 8 ch chip takes *a lot* more power and is certainly a design consideration. I wouldn’t be surprised if it’s fpga and not a dedicated adc chip, considering how multi-function the unit is
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Likely. I too suspect this is a cost-saving measure realized by halving the processing and powering requirements, regardless of the actual physical implementation.
Consider it to be somewhat analogous to the older non-auto-switching mode of recording a second set of safety tracks at lower level. Engaging that mode requires a second set of ADC channel paths and recorded output channels, but because no additional ADC>output-file channel paths are available, it cut the available recorded channel count in half. Very similar behavior.
Zoom and SD could have similarly limited the number of inputs when set to record 32-bit float mode, however the SD series II and Zoom F8 "Pro" generation recorders were designed with sufficient resources to be able to support "auto gain/ADC switching" at full input channel count.
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I was still toying with the idea of buying one of these, not because of 32-bit float but because of the form factor, 5V PiP and the hopes for a good preamp (which, I hear, is true of their Theos transmitters). But then I saw these were available:
https://store.lom.audio/products/usi-phantom-adapter?variant=4542168629280
Just ordered the adapter and a Zoom F3 instead. The phantom adapter runs on 24-48V phantom, and outputs 8V, so I'm hoping selecting 24V on the Zoom F3 will give me more than enough battery life for a single show on rechargeable AAs.
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Today I visited shitty gig to test PR2 in the wild )))
The sound in the club was pretty shitty for these type of shows :-\
For all mics gain was set to 0.
With DPA 4061 and Deity W.Lav Pro I recorded about 30 seconds in 32bit mono.
With CAFS I recorded almost whole show in 24bit stereo.
https://drive.google.com/drive/folders/1A0qMgdXZgy2mqee-9QrG4R2JiYKTf4ju?usp=share_link
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Today I visited shitty gig to test PR2 in the wild )))
The sound in the club was pretty shitty for these type of shows :-\
For all mics gain was set to 0.
With DPA 4061 and Deity W.Lav Pro I recorded about 30 seconds in 32bit mono.
With CAFS I recorded almost whole show in 24bit stereo.
https://drive.google.com/drive/folders/1A0qMgdXZgy2mqee-9QrG4R2JiYKTf4ju?usp=share_link
Thank you for the samples. That music is pretty intense, I love it.
Interesting results, were you standing in the same spot for all 3 samples? It sounds like the bass/ subwoofers are pretty strong for this show, and that 5v PiP is proving it's worth by the lack of distortion. Keep the samples coming if you can :coolguy:
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Thank you for the samples. That music is pretty intense, I love it.
Interesting results, were you standing in the same spot for all 3 samples? It sounds like the bass/ subwoofers are pretty strong for this show, and that 5v PiP is proving it's worth by the lack of distortion. Keep the samples coming if you can :coolguy:
Same spot all the time.
I standed in first row in the right side of the stage. It's not so good place for taping, but at rock shows at same spot results usually much better ::)
I hope i can visit punk show on saturday to record more samples >:D
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Hmmm. I have at last got around to reading the last few weeks of this thread. The whole "32 bit float is mono only" thing seems to be a bit terminal. I have a DJI Mic 2 which does that and is only a fraction of the size of the PR-2, and it has a built in mic as well as a lav mic input. And it transmits by wireless or bluetooth to suitable devices. How does it sound? Dunno, I have never got around to using it. But this whole 32 bit float space is now so packed with options for different purposes that the PR-2 has the feel of something that is out of date before it even properly hits the market. As for why it started off as stereo and now it's mono, that could be down to processor limitations having been discovered when it was too late to do anything about it. For instance, overheating issues.
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Gotham just told me deity pushed it back again, to mid july
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Gotham just told me deity pushed it back again, to mid july
But Psinka in Russia has one already. Deity is shipping to Putin but not anywhere else? I call for a Congressional investigation! Is this what we get for Lend-Lease!
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But Psinka in Russia has one already. Deity is shipping to Putin but not anywhere else? I call for a Congressional investigation! Is this what we get for Lend-Lease!
Hey! It's not a Deity fail and they didn't ship anything yet. It seems that problem in chinese suppliers that shipped small batch before they should.
When i bought mine i found one site that also got PR2 in stock - https://www.masterfoto.lv/en/sound-recorder/31880-1.html
For now they change status for pre-order.
PS: Putin is a sick crazy bastard who needs to be judged and jailed
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Deity on x just now…
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Gotham got their shipment today. I should have mine soon
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Now I just need something to record
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Now I just need something to record
Do you have english manual? ))) Mine came with chinese only (
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It came with a link to a manual that isn’t active, so no manual of any type yet. Took some googling to figure out how to turn off the lock I accidentally enabled, and I have no idea what the switch is for. Other than that, the app works well, and the always recording feature makes it easy. Just load everything in the hat at a safe level and go in
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Took some googling to figure out how to turn off the lock I accidentally enabled
HOW ??? I want to use lock but can't find how to turn it off :banging head:
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Took some googling to figure out how to turn off the lock I accidentally enabled
HOW ??? I want to use lock but can't find how to turn it off :banging head:
Triple click the power button
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Tiny. Looks like you could make some shortie cables and velcro the PR-2 directly to the center part of the mic-bar. Totally self contained in hat with no exposed wires.
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That's essentially the plan.
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Has the PIP voltage been confirmed yet?
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Has the PIP voltage been confirmed yet?
I assume it's the 5v in the specs, but I haven't put a meter to it.
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Has the PIP voltage been confirmed yet?
I assume it's the 5v in the specs, but I haven't put a meter to it.
Would be interested to see if it can power DPA 406x's safely and without any intermediate power supply. Would make for a super small 007 rig for places where you can hardly bring anything in.
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They just hit me with a price of $239 for the DR-2 (US)
US model doesn't have MIC pass-through. (artifact of DMCA?)
Manual here:
https://deitymic.com/wp-content/uploads/2024/07/SRD-mini-EN.pdf
Methinks it may be manufactured in China
So, I Velcro this on top of my Mini-ME? ^-^
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They just hit me with a price of $239 for the DR-2 (US)
US model doesn't have MIC pass-through. (artifact of DMCA?)
Manual here:
https://deitymic.com/wp-content/uploads/2024/07/SRD-mini-EN.pdf
That's the wrong manual, even tho it's what deity links to from the pr2 page.
Methinks it may be manufactured in China
well, sure
So, I Velcro this on top of my Mini-ME? ^-^
To what end? It doesn't have spdif/aes input, and won't pass audio signal when recording (because of the Zaxcom patent).
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They also relase a firmware update ::) https://deitymic.com/firmware/
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can you flash the non-US firmware (not sure its even downloadable yet) to get around the limitations?
losing the ability to monitor puts it behind other small recorders in my use case
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They also relase a firmware update ::) https://deitymic.com/firmware/
looks like one file. it will either autodetect the us version and leave it crippled (my guess), or it will flash all to be full featured
perhaps someone can hex-edit the .bin in the former case (at risk of bricking the device of course)
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They also relase a firmware update ::) https://deitymic.com/firmware/
does your unit pass signal when recording? or did you get a US model?
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does your unit pass signal when recording? or did you get a US model?
I got non US model 100% )
I can check signal passing when return home in about 4-5 hours
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I got non US model 100% )
how are they distinguished?
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That's the wrong manual, even tho it's what deity links to from the pr2 page.
On product page for PR-2 US https://deitymic.com/products/pr-2-us/ we cqan find correct manual https://deitymic.com/wp-content/uploads/2024/07/PR-2-manual-EN.pdf
UPD: they fixed links on booth pages)
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If this had floating 32-bit in stereo I would get it in a heartbeat. But since it doesn't, there's no way I would get it
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Strange, the link that I saved was correct, I think I fudged it in the copy/paste... My bad
I'd like to hear something 32bit from it!
And a few solid opinions
I was sort of joking about the Velcro, and I assume everything good would come in post, after a 32bit capture.
So any good powered mic set in, and the magic in the DAW.
Ultimately better in that preamp modeling allows a pivot to the best/preferred sound.
I doubt it is brickable, most modern electronics will boot-load from media that is formated and configured properly.
My worry would be in obtaining that from support, and how strong communication will be.
They certainly have solutions in mind
Speaking of preamp modeling....
Has anyone put out filters that represent the Neve, Apogee, SoundDevices family for "coloration"
Obviously, purity cannot be synthesized....
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I'm looking at the Deity recorder, not for concert recording but for interviews and so on. I've downloaded the manual (unfortunately way too large a file to attach here) and will read it through before deciding. I'd need to get a new pair of microphones to use solely with it, which I don't look forward to, or else work out some sort of adapter for a pair of Schoeps capsules on active cables. All in all, it's at most a maybe.
-- I don't want to be tedious and pedantic, but in this this thread many people have mentioned 32-bit recording as being an attraction of this type of recorder. I think they may misunderstand what that feature can and can't give you. Just to keep it brief, Deity's claim:
> 32-bit float means you’ll always have perfect gain staging
... is misleading and wrong, but an interesting choice of misstatement since it caters to the illusion that many people seem to have. I've extracted the specs from Deity's manual; they're attached below. Please note the s/n spec, which is about that of a typical 16-bit recorder. Note that even that spec is achieved only if the preamp gain is set to a rather unlikely value.
"32-bit float" is a data format--a way of storing the sample values that come out of an A/D converter. That converter will have a certain noise floor and maximum output level given the circuit it is in; the best available converters can give ~22 bits of range between that floor and that ceiling. The 32-bit encoder then takes this (less-than-24-bit) output from the converter and translates the linear PCM samples into a floating-point binary format for storage. But no encoding of those sample values, no matter how many bits it uses, can possibly have any greater dynamic range than the A/D converter that provided them. Thus the fact that the 32-bit float data format (in the abstract) has its own, much wider dynamic range is irrelevant.
For analog microphones, any digital recorder has to have an analog mike preamp followed by an A/D converter. A given pair of microphones in a given recording situation will produce a noise floor and a peak output voltage; both will vary according to the sensitivity and noise of the microphones, and the loudest and softest sounds that occur. There's a huge amount of variation among those things--no one preset gain for a mike preamp can possibly cover all situations optimally. If that gain is too low, the input noise of the preamp may exceed the noise contained in the signal from the microphones; if it's too high, overload can occur in the preamp and/or the A/D. The fact that the A/D's output is then translated into a 32-bit data format has NO EFFECT on that situation, since it comes into play only once the numeric sample values come out of the A/D converter.
What's required, ironically, is proper gain staging--the thing that the ads say is guaranteed to be "perfect" because of the 32-bit data format, but in fact isn't guaranteed at all.
[Edited later to add: The lack of a "pass-through" feature in the U.S. version of this product means that once you start recording, you can't hear anything through the headphone jack. It reminds me of two-head versus three-head analog tape recorders, except that this is like zero-head; the meters move, but you don't know what you're getting. I don't know whether that cutoff occurs in pause mode or not. If it does, then I'm no longer interested in this recorder at all, since to set up for recording you'd have to record a test, then play it back separately; there'd be no adjusting the microphone position while listening in real time.]
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^ As far as I know, the 32-bit float "no gain staging needed" thing is actually a product of using multiple gain-ranged ADCs, operating on overlapping bands of input, and stitched back together using regression based algorithms. I believe this to be true, at least, for the SD MixPre series. The 32-bit FP "bucket" is useful for this, because it creates a broad range into which the (max) 24-bit recording can occupy. My 2 cents and all...
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all valid poitns dsatz. i too am off the wagon for a US version as my use case is stealth and im always actively monitoring so i can dodge the talkers. given the cost, if i had a use for this i would seek out a grey model non-us version, warranty be damned.
32-bit is the last feature i look for in a recorder, i rarely use it in the 32 bit devices i have now unless it is 100% remote/unattended. Your read of the specs is correct, i saw some RMAA measurement of this device and its in the low 80s in dynamic range, and again thats at optimum line-in level staging*. 24 bit word is more than enough to contain its resolution on its best day
*the tester was using a source with +6dBU max out. assumedly the specified dynamic range is at +18dbU input. My best guess with a mic in signal based on the limited data i have is a dynamic range in the upper 60s to mid 70s, certainly nothing approaching state-of-the-art
id fully test it but im leery to drop $250 on something i probably wouldnt be keeping
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The scheme by which a recorder handles the input signal through the ADC is one thing. The file format that is saved is another entirely different thing.
32bit float is the output file format. That alone tells us very little about the recorder. The industry choosing to use "32-bit float recording" as the way of describing and marketing this new class of digital recorders is quite unfortunate, and in my opinion verges on misleading. What happens ahead of the file format is the important part and what should really be better explained by the manufacturers. A good name for that is how this class of recorders should be described. Maybe extended input range functionality or something. Others will come up with better descriptions I'm sure.
All recorders using this approach essentially do the same thing by automatically switching between several separate internal signal chains, each of which is optimized for a particular range of signal levels. Each path is essentially a separate preamp > ADC channel. Prior to the internal preamp>ADC channel thats internally gain-staged to provide best low-level performance overloads, the recorder switches to using the output from a separate internal preamp>ADC channel which has been gain-staged to handle higher level signals. The output is stitched back together (which is the tricky part) before being written to the output file format. The written output format might be 32-bit float, 24bit fixed, MP3 or whatever.
In this particular recorder, I suspect the reason it is only able to record a single channel when using the extended input range functionality is because there are only two preamp>ADC paths available. When in stereo mode each input channel uses one signal path, when in mono mode the single input channel uses both.
I suspect the 90dB s/n spec reflects each individual preamp>ADC channel, while the EIN -130dBu (A-weighting +30dB @ 150ohm) spec, and the Dynamic Range 123dB typ(32-bit Float) spec, reflects the the combined extended input range made available by switching between both 90dB S/N input>ADC circuits, and stitching their outputs back together again.
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I might pick one up just to have an easy way for a friend to record something while I'm away in Sept. But for my own more dedicated use I'm awaiting a good technical review with measurements of stereo mode performance, a few recorded music audio samples, and confirmation that multiple units can be: 1) reliably controlled simultaneously from a phone, and 2) clock sync'd to a sufficient degree of precision. I expect I'll probably need to confirm 2) myself.
I don't really care about the 32-bit float feature, as long as performance in stereo mode meets or exceeds that of the Tascam DR2d in 24/48. Unfortunately, as discussed above, dynamic range performance in stereo mode is not clearly specified, so I await actual measurements. If it pans out, I'll be in for two of these to replace 4ch-CA-UGLY> DR2d, with the potential to expand to 3 or 4.
The info Deity has posted about remote control of multiple units via the Sidus Audio app looks good, but I need to contact them to ask more about what is required for clock-sync. Hoping a phone running the app connecting to the PR-2s is all that will be required, but I'm guessing I may also need a Deity TC-1 wireless timecode generator. TC-1 is small, inexpensive, and wireless, but I rather not deal with and additional device in pocket and its batteries.
Working my current rig down to 2 small, dedicated audio devices (4ch_CA-Ugly > DR2d) was a challenge that took years. This is the first new setup that may be capable of practically replacing it. If TC-1 is not required, the dedicated audio device in pocket count remains the same, while the two devices are significantly smaller and easier to stealth (I'm not counting the phone as a dedicated audio device as it will only be used for control, not storage, and it gets carried along either way).
If I need TC-1 to sync multiple units effectively, that will require carrying an additional (but smaller) box, but at least no additional wires which is the bigger hassle. Recording 6 channels would need four dedicated audio devices, yet they would collectively take up about the same pocket space as my current 4ch rig, and would probably be a better form factor.
Lack of pass-through in the US version is unfortunate, yet not really a deal killer for my use. Its helpful when trouble-shooting, but use primarily entails recording without monitoring and assessing the results later.
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I'm sure that I've seen this here before, but:
https://www.sounddevices.com/32-bit-float-files-explained/
More dynamic range than the moon to the earth....
I agree, the analogue Preamps won't get much better than 80 - 90 DB in any $250 product
And even that is generous, but all I need is about 40 - 50 DB of headroom for a really smashing live recording.
I'd lie if I didn't say that I sometimes normalize when live mastering, after NR.
Sometimes it just sounds better loud
In a dead-quiet studio, life and settings are different
And, will a $250 box ever even approach the SD method and algorithms ?
"Float" is the key, 32bits is easy, finding "zero" the "art and soul" of the machine.
'art... punning'
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I don't really care about the 32-bit float feature, as long as performance in stereo mode meets or exceeds that of the Tascam DR2d in 24/48.
that would be a bar for sure. the DR2d is one of the cleaner handhelds out there and tops its sony and edirol counterparts in dynamic range when measured
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aaronji wrote:
> ^ As far as I know, the 32-bit float "no gain staging needed" thing is actually a product of using multiple gain-ranged ADCs, operating on overlapping bands of input, and stitched back together using regression based algorithms.
That would work only if multiple, gain-ranged mike preamps were linked to those multiple gain-ranged ADCs. It's not impossible, but I would never simply assume that a recorder (especially a low-cost one) has such a feature. Otherwise the gain of the mike preamp needs to be user-adjustable, and the user needs to take care to set it more or less optimally, just as with conventional record levels. If a recorder doesn't have one or the other, and users are crowing about how it never overloads, then you can pretty solidly infer that some users are making noisier recordings than necessary, despite however many bits the audio (plus preamp noise) is being recorded with.
[edited later to point out:] If a recorder had the kind of gain-ranging stuff built in that people here seem to imagine that they do, they would have signal-to-noise ratios on the order of 120 dB or better, measured analog in to digital out. But they don't. Similarly, if that type of technology were readily available at low cost, why don't we see a new generation of combination mike preamps / A/D converter units with signal-to-noise ratios several orders of magnitude better than anything that's been on the market before? They would easily outperform all existing separate preamps and converters, and all existing combination units of the older generation (Lunatec V3, Sound Devices MixPre-D, etc.) by tens of decibels. All older equipment would be massively obsolete overnight--and if that's happened, I'm not aware of it.
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aaronji wrote:
> ^ As far as I know, the 32-bit float "no gain staging needed" thing is actually a product of using multiple gain-ranged ADCs, operating on overlapping bands of input, and stitched back together using regression based algorithms.
That would work only if multiple, gain-ranged mike preamps were linked to those multiple gain-ranged ADCs. It's not impossible, but I would never simply assume that a recorder (especially a low-cost one) has such a feature.
Otherwise the gain of the mike preamp needs to be user-adjustable, and the user needs to take care to set it more or less optimally, just as with conventional record levels.
its more than likely just a single-ranging ADC with a simple IC gain circuit that has the <100 dB available to it well within the 144 dB 24-bit container, and 32 bit file format is just useless window dressing. but wont know until i try one. like all devices these have maximum input levels that can be exceeded. if the max input really is +18 dB (crazy high for an unbalanced signal) then it indeed has some headroom to work with, albeit at the expense of a higher noise floor for those hoping to use it as a mic-in
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That would work only if multiple, gain-ranged mike preamps were linked to those multiple gain-ranged ADCs. It's not impossible, but I would never simply assume that a recorder (especially a low-cost one) has such a feature.
I don't understand why you say there needs to be multiple preamp circuits. It's just one preamp with fixed gain, and the signal is split to the two or three parallel ADCs. Aaronji is correctly describing how Sound Devices, Zoom, and (I'm pretty sure) Tascam, and Stagetec do it.
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I see one difference in particular with regards to the architecture of this recorder when compared to the SoundDevices and Zoom recorders, but that difference is not the one we are discussing.
I think by design, all of these recorders use separate analog circuit paths ahead of each separate ADC path. I don't see how they could otherwise provide the dynamic range that most of them do without some kind of strategy that actively adjusted the preamp gain. Rather, each analog circuit > ADC path is optimized and 'locked down in terms of gain-staging' for its particular voltage range. That strategy provides a way around both dynamic range bottle necks: That of a single analog input path ahead of the ADC and that of a single ADC path itself. Otherwise in an SD Mixpre II, a single analog preamp path without any gain adjustment feeding multiple ADC paths for each channel would have to be capable of providing SD's claim of "142 dB of dynamic range" minimum on its own. That's extreme! I'm no expert but that level of performance may not be possible from a single fixed-gain preamp stage at any price point.
For that reason I don't see how any of these recorders would work to specification without using multiple preamp paths feeding the multiple ADC paths, with each optimized for a particular range.
Here is the difference I see with the PR-2:
Regardless of which recording mode is chosen, the SD and Zoom recorders provide a sufficient number of parallel signal paths to support their total channel count. When we were discussing the F6 noise-floor modulation issue that cropped up a while back, it was confirmed that these machines always use their extended-range ADC switching strategy regardless of which output file format the user selects. I'd originally assumed that when using those recorders in 24bit-fixed mode they would revert to a traditional circuit path arrangement, but apparently that is not the case. The SD and Zoom recorders always do the auto-ADC-switching thing, even when set to save the output as 16 or 24bit. Its the only preamp>ADC architecture they use, regardless of how many channels are being used or output format.
The PR-2 appears to use a different strategy. It presumably only has two signal paths available to it, so it doesn't have sufficient parallel signal paths to support a full channel count regardless of recording mode. Because of that it needs to use a different strategy ahead of the section that does the "digital switching and stitching" prior to the output file being written. The two available signal paths can either be used separately in a traditional way for 2-channel throughput (stereo), or used together in a "switching and stitching" parallel mode to provide increased dynamic range for a single channel throughput (mono). In mono mode, provision for the selection of output file formats other than 32-bit float could be made but isn't, which is simply a design choice. So unlike the SD and Zoom recorders, the PR-2 uses two different path architectures, switching between "traditional stereo mode" and "auto-ADC-switching mono mode" as a way of maximizing the utility of having just two signal paths available to it.
That strategy reduces cost by eliminating the additional parallel paths that would otherwise be required to provide increased dynamic range functionality for two channels.
In the SD and Zoom recorders we are never exposed to nor aware of the individual paths. They are always stitched back together again prior to output. But with the PR-2 in stereo mode, we are using the individual paths separately. Diety must set up the mono channel switching architecture in such a way that when mono 32-bit mode is selected it places the two channels in parallel and increases preamp gain through one of the two to appropriately gain-stage it for use as the lower level path. And after that the stitching back together and file writing takes place.
What is stated in the PR-2 specs is the dynamic range of the two paths used together in mono channel mode. What remains unstated in the specs is the dynamic range through each of the two individual signal paths.
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As I understand it, after the analog preamp, each ADC has its own analog gain block and filter, at least in SD's implementation. That's based on a brief conversation with an SD tech...
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Okay thanks, I accept that. Can anyone here more electrically astute than myself comment on how difficult (or not) it is to achieve a dynamic range of >142dB through a fixed gain preamp? Am I interpreting that achievement as being more of a reach than it actually is?
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Okay thanks, I accept that. Can anyone here more electrically astute than myself comment on how difficult (or not) it is to achieve a dynamic range of >142dB through a fixed gain preamp? Am I interpreting that achievement as being more of a reach than it actually is?
sounds impossible as EIN gets amplified when you amplify any signal, you need to jigger it a bit and combine multiple signal paths
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The FLOPS are in the maths (https://en.m.wikipedia.org/wiki/Single-precision_floating-point_format).
The ADC needs multiple paths. (https://www.analog.com/en/products/ltc2508-32.html)
The preamp is before them.
Math and integrated circuits? I abhor them.
I'm going to the beach! (https://www.wired.com/story/32-bit-float-audio-explained/)
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... you need to jigger it a bit...
Love the technical jargon, my kind of language! :yack:
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Newsshooter have posted a short review.
https://www.newsshooter.com/2024/07/18/review-of-the-deity-pr-2-pocket-recorder-with-32-bit-float-and-timecode-generator/
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Thanks for that link, from which I quote
-"It can’t fix improper mic placement that will distort the microphone, but the recording level can be adjusted a lot in post, just as long as you don’t blow out the microphone. "
That kind of sums up the whole subject of 32 bit float in one simple sentence. Though perhaps one could add "or preamp".
It seems to me that the best use of 32 bit float is for the implementation of mics that record. The design of such devices can ensure that the mic can handle high levels and that the preamp is optimised for the mic, possibly including corrective eq if necessary. That can then be passed on to the dual a/d converters and recorded internally without cable losses in 32 bit float. Such mics exist but how well the theory matches the reality is another whole matter, probably dictated by the price point.
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I dunno. For me, the best use of this technology is the ability to record gigs without having to check levels as the music rises and falls in volume in ways that cannot be predicted ahead of time. The stealth utility is obvious. In open situations, it means one can set up more than one rig at optimal locations on or near the stage that are not accessible to the taper when the music is playing. The F3's pre-amps are pretty darn good and the recorder is a good match for Line Audio CM3s.
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I dunno. For me, the best use of this technology is the ability to record gigs without having to check levels as the music rises and falls in volume in ways that cannot be predicted ahead of time. The stealth utility is obvious. In open situations, it means one can set up more than one rig at optimal locations on or near the stage that are not accessible to the taper when the music is playing. The F3's pre-amps are pretty darn good and the recorder is a good match for Line Audio CM3s.
Yeah, I agree, but I think their point is that the recorder's manufacturer could match the analogue clipping point of the recorder to 0 dBS, and then configure multiple ADCs to write to a certain volume range of a 24 bit file. For this to work, you wouldn't be given the option of gain control either. You'd never clip digitally (unless the analogue stage also clipped), and the dynamic range of 24 bit is said to be enough. But my question is... why? The difference in terms of storage is not huge, and your quietest sounds would still be written at -50, -60, -70dB in the file and demand normalisation. Using 32-bit float is a pretty foolproof way of making sure the user will never run into quantization noise or accidentally clip the audio during post-production.
It's also just crossed my mind that distorting the analogue stage sounds very different from digital clipping. Wouldn't you need more digital headroom there to capture the analogue distortion? And if you do, wouldn't you be writing the quietest sounds closer and closer to the quantization noise? And we must remember: just because a sound isn't completely drowned out by quantization noise, that doesn't mean you can't hear it in the background. At which point I'd ask: isn't it just easier to move on to 32-bit float? There are no downsides other than using a little more storage.
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I still think it's pretty silly to have auto ranging ADCs with 32-bit float on a device with rather poor S/N and dynamic range numbers by modern standards. Just as silly as Zoom doing it with the H Essentials line. True, you get the benefit of no level setting. But if the device is going to be built on this architecture, you should also be getting the benefit of increased dynamic range.
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I still think it's pretty silly to have auto ranging ADCs with 32-bit float on a device with rather poor S/N and dynamic range numbers by modern standards. Just as silly as Zoom doing it with the H Essentials line. True, you get the benefit of no level setting. But if the device is going to be built on this architecture, you should also be getting the benefit of increased dynamic range.
Yeah, I agree with you there re: the H Essentials line, especially since it appears to have worse S/N than the H and HxN lines. That's why I didn't buy one of those. I actually used the Zoom H1 for a long time with a Church Audio pre-amp, and I had no issues whatsoever with noise when the music's dynamic range wasn't too great. A song at -20 dB on average with the occasional -6 dB peak? No issues. But as soon as the band got really quiet at -30 dB or less, the noise was fairly noticeable. Increasing the gain on the Church Audio pre-amp would have fixed the issue, but then the peaks would clip.
But is the Deity's mono 32-bit mode as bad as the Zoom H Essential's? I've never seen a rigorous analysis out there, but I heard their mono 32-bit transmitter (the Theos) was really good - one youtuber said it was 'professional' quality, comparable to Sound Devices. I assumed the PR-2's circuits would be essentially the same, but with the option to record 24-bit stereo as well.
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I still think it's pretty silly to have auto ranging ADCs with 32-bit float on a device with rather poor S/N and dynamic range numbers by modern standards. Just as silly as Zoom doing it with the H Essentials line. True, you get the benefit of no level setting. But if the device is going to be built on this architecture, you should also be getting the benefit of increased dynamic range.
I see your point, but I guess the thing is that it seems that 32 bit float chips are cheap now, and we might as well get used to that as the new norm. I look at it this way - dynamic range and distortion are affectd by the mic, the preamp, and the a/d conversion. If one of these three at least can be taken off that list, it helps. But the danger is that people may start to overlook the part played by the first two. Today I came across a video by a college lecturer in digital audio who said in commenting on a post questioning the preamps of the H1e that any noise problems from the preamps would be dealt with by the 32 bit float format. Of course I pointed out that those a/d converters would make a very good recording of that noise!
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But the danger is that people may start to overlook the part played by the first two.
Absolutely, and that's what some of the manufacturers implementing this are leading people to believe.
Today I came across a video by a college lecturer in digital audio who said in commenting on a post questioning the preamps of the H1e that any noise problems from the preamps would be dealt with by the 32 bit float format. Of course I pointed out that those a/d converters would make a very good recording of that noise!
It sounds like this person is not well qualified for their job.
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^^ Depends on the source of the noise, but agreed that it is unlikely.
FLoating point OPerationS (FLOPS) gives massive Dynamic Range (DR), and is not a panacea.
It is the foundation of an easy to control digital recorder.
Most say it eliminates the need to set levels, the rest is from other parts in design.
Doubled input of differential levels are mathematically conjoined so quiet sounds and loud sounds are more faithfully captured.
This occurs internally, and after the microphone or input connections.
I notice that discussion of "coloration" in the preamp stages has been totally abandoned.
Transparency, I would assume, is now completely the goal.
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I got my hands on a PR-2, trying it out to find out if there are any dealbreakers…
So far i have come across one thing that is far from ideal, in my opinion (i only do stealth)…
When recording in stereo mode, which i would always do, the sidus app only has separate gain for left and right, and the gain interface is based on sliders… already i have accidentally hit one of them on changed the gain significantly on one of the channels.
Is there any way to sync the gain on both channels, and ideally have up/down buttons instead? Some setting i haven’t seen…?
Unless i find a way around this, i willl have to return the pr-2 and stick to my a10…
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^ Err! regarding simple stuff like that that ends up foiling good plans. If there is not a way to do it, maybe with enough feedback they'll update the app with a way to lock down gains, or shift those controls to a separate page. I hoped to use the Zoom F8 app just to start/stop recording and to monitor meters. But I don't because a few things similar to that make it effectively unusable for me.
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Most say it eliminates the need to set levels, the rest is from other parts in design.
Doubled input of differential levels are mathematically conjoined so quiet sounds and loud sounds are more faithfully captured.
This occurs internally, and after the microphone or input connections.
Second bold first- Not more faithfully captured. Sampling theory.
First bold- Level adjustment is unavoidable. It just makes it possible to shift the making of that adjustment until afterward in many but not all cases. It can do so as long as the signal fits within the input S/N of the recorder, which is the real-world bottle neck, not the output format. Recorder and interface implementations using multiple auto-switched ADCs ease that real-world bottle neck to a desirable practical degree. That's the selling point and why people like it.. and yet the same bit-identical output could also be handled by a 24bit fixed-point output file, in which case the recorded material would require the same level adjustment made to it afterward.
I'm going to the beach! (https://www.wired.com/story/32-bit-float-audio-explained/)
^That linked Wired article on "32-bit-float-audio-explained" is shamefully weak. This statement in particular is just plain flat out wrong and makes it clear the author has no idea of what he's writing about-
"Now, typically you’ll set audio levels when setting up your equipment to avoid hitting that limit. Setting those levels involves applying gain to the signal from the mic, which is an irreversible step that crushes the dynamic range of even 24-bit recording."
But your link title makes for a good lead-in to a quick story about my bike ride over to the beach this past weekend. As I was up out of the saddle of a single-speed beach-cruiser cresting the apex of the barrier-island draw-bride, some guy walking along the sidewalk with his friends held out a bottle of water for me. I grabbed it and immediately poured it over my head while shouting, "Yellow Jersey Tomorrow!" Half the folks on the sidewalk roared in laughter, while the other half just looked confused. C'est la vie!
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^ Err! regarding simple stuff like that that ends up foiling good plans. If there is not a way to do it, maybe with enough feedback they'll update the app with a way to lock down gains, or shift those controls to a separate page. I hoped to use the Zoom F8 app just to start/stop recording and to monitor meters. But I don't because a few things similar to that make it effectively unusable for me.
My plan is to set the levels, conservatively knowing the microphones and the genre so that I don’t have to do on the fly adjustments for levels. 24 bits and a 90 dB signal. The noise should leave plenty of room for that in most low profile situations.
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That's my plan as well, and how I currently use DR2d. It wouldn't be a concern running a single PR-2, in which case I could just forgo using the app in the same way that I run the F8. My plan while recording is to use the app only for start/stop transport control and clock sync of twos or more PR-2, and simple confirmation everything is working.
The concern raised by colargol's post is the potential for making accidental gain adjustment while doing that.
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I don’t personally see a lot of risk in that.
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Probably just need to download the app and play around with it in demo mode to determine if my fingers are overly fat or not.
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Demo mode doesn’t currently show the pr2
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I sent some feedback to the Sidus audio app people, basically suggesting a sync setting for the left/right gain adjustment options, and also an option to use up/down buttons instead of a gain slider. I hope they will consider it…
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Curtis Judd uploaded his review yesterday:
https://youtu.be/BQTBaxsGbL0?si=LkgrfAA4O7H59x3g (https://youtu.be/BQTBaxsGbL0?si=LkgrfAA4O7H59x3g)
He represents the intended customer (film makers), but you may extract something anyway.
His reviews are typically very sober, adult and to the point - rarely longer than they have to be, but if you just want the conclusion, you can skip to 12:19 for lists of his perceived Pros and Cons. https://youtu.be/BQTBaxsGbL0?si=E1owbEGpOVh-yxSL&t=739 (https://youtu.be/BQTBaxsGbL0?si=E1owbEGpOVh-yxSL&t=739)
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No 32-bit Float Stereo, likely because it usurps the 2nd ADC for the float function.
In stereo it uses one per channel, I am guessing...
Thanks for the link!
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I suspect the 90dB s/n spec reflects each individual preamp>ADC channel, while the EIN -130dBu (A-weighting +30dB @ 150ohm) spec, and the Dynamic Range 123dB typ(32-bit Float) spec, reflects the the combined extended input range made available by switching between both 90dB S/N input>ADC circuits, and stitching their outputs back together again.
Actually, the -130 dBu EIN looks a bit dubious. Let's compare this number to some other recorders. The MixPre-6 has an a-weighted EIN of -128 dBU, Zoom F8n has -127 dBu. I doubt the Deity recorder has better noise performance than both of these recorders. In fact, the stated -130 dBu EIN would put the Deity in the same league as i.e. Sonosax. The 150 Ω source resistance (which they claim they used to load the input) emits around -131 dBu of noise at room temperature. And the Deity's EIN is supposed to be just a decibel higher?
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Yes, good point and I agree. Thanks for quantifying it. To be clear, my speculation was about what those specs are measuring, not their accuracy.
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Yes, good point and I agree. Thanks for quantifying it. To be clear, my speculation was about what those specs are measuring, not their accuracy.
Yes, I know what you meant. You mentioning the EIN just got me thinking about the validity of that spec sheet, that's all. I hadn't even looked at those numbers before because I'm not really in the market for another recorder right now.
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Is 5 volts pluginpower enough to skip battery box completely?
I currently have ChurchAudio CA-14 microphones.
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I don’t know about the ca mics but it’s definitely enough for dpas
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PR-2 just arrived. It's downright diminutive!
Plan to test run it at an amphitheater show this coming Mon night with a pair of DPA 4060 CORE. Will play around with it a bit first of course. Anyone record any music yet using 4060 or 4061? Wondering about input gain setting, and presume 0dBu will be about right.
Other than the test run next Monday night, my initial goal is an easy to use stealth rig I can pass to a friend to run for a show in September while I'm out of town.
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PR-2 just arrived. It's downright diminutive!
Plan to test run it at an amphitheater show this coming Mon night with a pair of DPA 4060 CORE. Will play around with it a bit first of course. Anyone record any music yet using 4060 or 4061? Wondering about input gain setting, and presume 0dBu will be about right.
Other than the test run next Monday night, my initial goal is an easy to use stealth rig I can pass to a friend to run for a show in September while I'm out of town.
Gut how are you dpas terminated ? is there a micro to locking stereo 1/8" Y cable ?
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PR-2 just arrived. It's downright diminutive!
Plan to test run it at an amphitheater show this coming Mon night with a pair of DPA 4060 CORE. Will play around with it a bit first of course. Anyone record any music yet using 4060 or 4061? Wondering about input gain setting, and presume 0dBu will be about right.
Other than the test run next Monday night, my initial goal is an easy to use stealth rig I can pass to a friend to run for a show in September while I'm out of town.
Gut how are you dpas terminated ? is there a micro to locking stereo 1/8" Y cable ?
I bought a microdot to locking 1/8” connector. I’ll try it out when the pr2 gets back to me (loaned it to a friend for testing)
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PR-2 just arrived. It's downright diminutive!
Plan to test run it at an amphitheater show this coming Mon night with a pair of DPA 4060 CORE. Will play around with it a bit first of course. Anyone record any music yet using 4060 or 4061? Wondering about input gain setting, and presume 0dBu will be about right.
Other than the test run next Monday night, my initial goal is an easy to use stealth rig I can pass to a friend to run for a show in September while I'm out of town.
Gut how are you dpas terminated ? is there a micro to locking stereo 1/8" Y cable ?
I bought a microdot to locking 1/8” connector. I’ll try it out when the pr2 gets back to me (loaned it to a friend for testing)
neat
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Mircodot. This Monday I'll use one of a handful of DIY 2Xmicrodot>stereo-miniplug Y adapters I made many years ago (non-locking). But just ordered a locking stereo-mini Y off ebay for $11, due to arrive before I depart in September, which I plan to use instead for the newb taper friend I'm sending in my place, just to eliminate that potential point of failure.
Monday amphitheater show is a Lee Fields (soul) opener / Charlie Crockett (country) main act.
Upcoming theater show in Sept is BEAT (Belew, Vai, Levin, Carey) doing 80's King Crimson.
I could alternately run 4061 legacy if a better match. 4060 is fully capable of handling the SPL of these shows, but I'm not sure what the optimal gain staging with the PR2 will be yet.
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Excited to hear how the DPA > PR2 setup works!
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DPA 4060 > Deity PR2 would be microscopically small. There's a certain elegance to how small this rig is.
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I have found after 4 shows and about 12 or 13 hours of recording...
that the metering of the PR-2 is somewhere around 5dB COLDER than actual
when reading the display the bars are WHITE until about -16dB
when they change to YELLOW until about -10dB
after that they go to RED up to 0dB
I am shooting for peaks around -10dB now
and get peaks around -5dB when opening up with Amadeus Pro on a Mac
I set my input to -3dB using an AERCO MP-2 running BOTH the PR-2 and a Sony A10
I use the A10 to judge the input on the PR-2 relevant to the AERCO and as a backup
the A10 is running at 3 (or 2 if I'm really close the source) on the input which does not leave a lot of headroom
having the ability to go down to -12dB on the PR-2 is nice
I also ran it once with a Baby NBox recording close to the source
and think running about +9dB was about right to avoid go over -10dB on the display
I think in MOST situations I would go +12 or even 15dB
I know my tests are not relevant to most PR-2 users here and am going to try it with some DPA 4061s when I get the "Y" cord
EDIT:
I am using Energizer Ultimate Lithium "AA"s and the battery display still reads FULL after 13 hours
I like the form factor and find the menu fairly simple to navigate
it has a 4 second countdown on the RECORD button for both starting and stopping to avoid accidents
I DO NOT use the 3 click LOCK feature
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I have found after 4 shows and about 12 or 13 hours of recording...
that the metering of the PR-2 is somewhere around 5dB COLDER than actual
when reading the display the bars are WHITE until about -16dB
when they change to YELLOW until about -10dB
after that they go to RED up to 0dB
I am shooting for peaks around -10dB now
and get peaks around -5dB when opening up with Amadeus Pro on a Mac
I set my input to -3dB using an AERCO MP-2 running BOTH the PR-2 and a Sony A10
I use the A10 to judge the input on the PR-2 relevant to the AERCO and as a backup
the A10 is running at 3 (or 2 if I'm really close the source) on the input which does not leave a lot of headroom
having the ability to go down to -12dB on the PR-2 is nice
I also ran it once with a Baby NBox recording close to the source
and think running about +9dB was about right to avoid go over -10dB on the display
I think in MOST situations I would go +12 or even 15dB
I know my tests are not relevant to most PR-2 users here and am going to try it with some DPA 4061s when I get the "Y" cord
EDIT:
I am using Energizer Ultimate Lithium "AA"s and the battery display still reads FULL after 13 hours
I like the form factor and find the menu fairly simple to navigate
it has a 4 second countdown on the RECORD button for both starting and stopping to avoid accidents
I DO NOT use the 3 click LOCK feature
Nice!!
For me, I am impressed with the PIP. I have run CA 11, CA 14 and AT 853's without the usual preamp, and the sound is really excellent. I do use the 3 click lock and like it . Battery life is fantastic so no need for an external battery. The screen really is easy to read, and the app works great although I do not rely on it. Great setup for any "challenging" venues as it is super compact. I do not see using the 32bit float sadly as it is mono, but the convenience and portability make this a great go to right now.
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Nice. We're starting to get some real usage data. My brief initial assessment last night parallels darby's and dallman's comments.
the metering of the PR-2 is somewhere around 5dB COLDER than actual
That probably reflects the metering being VU rather than a peak meter. Would be preferable if the PR2 displayed both as is common on other gear, combining the VU solid bar with a peak-line that has some some momentary held delay to it.
For folks who may not be aware, VU metering is "averaged" with a 300ms rise time that makes it more reflective of human hearing and traditional analog meters, but it does not indicate short transient peaks. The actual difference in indicated level between what a UV and peak meter display is going to vary with the nature of the signal and how fast it changes. In actual use, like Darby mentions, we'll just need to get a good feel for where we want the VU meter to top out, along with the knowledge that the actual peaks (dBfs) will be higher.. How much higher depends on how dynamic the music is. When recording on stage near the drum kit the peaks will be significantly higher than what the VU meter indicates, while when recording from far in back where its way less "peaky", the difference between the two will be significantly smaller. With an unchanging sine-wave input, both UV and a peak meter should indicate the same level.
In my initial playing around with the recorder last night, it seems easy enough to use. A long press of the record wheel button (along with a 3 second count down indicator on the display) starts or stops recording, unless the unit is set to start recording immediately upon power up, or upon wireless timecode sync.
A quick press of the record wheel button enters the menu, and the entry point is the input gain setting. If not locked, tapers will need to be careful not to accidentally double [edit] triple press the record wheel while recording (with no time limit imposed between presses) as the second press selects GAIN, and the third selects the LEFT channel for gain adjustment (when in stereo mode). Because there is no time limitation between those button presses, it would not be that unlikely to happen in pocket, and any inadvertent turn of the record wheel button after that will change the left channel gain setting. To exit the menu requires a single quick press of the power button which doubles as the "back". For that reason stealthers will want to engage the hold function which can be engaged manually, or set to engage automatically after 15sec or 1min. A triple click of the power button (in quick succession) locks/unlocks hold.
Similarly to what colagol mentioned about the Sidus control app, there is currently no way to link the gain adjustment of the two channels. Left and right channel gain must be adjusted separately.
IMO, neither the metering, the unlinked gains, nor the danger of accidentally changing the left channel gain are deal killers. Like others have mentioned, I plan to figure out what gain to run based on the mics I'm using, pretty much leave the gain untouched from there, and engage hold as soon as I start recording. However, it would be nice if a future firmware update addressed those things and I see no reason why they could not be updated via firmware.
The 5.6.5 Power supply adjustment of microphone section of the user manual states:
This mode allows you to manually switch the driving voltage of the microphone and select the microphone input option according to the type you need.
However, in the actual menu I find no option for switching the mic powering voltage. It just allows me to switch between line-in / mic-in. I've not yet measured the mic powering voltage. So far, I've only played around with it using the included mic. Will need to dig out a microdot Y adapter to try it with the DPAs.
[edit to clarify that 3 subsequent clicks of the record wheel, without any time restriction between clicks, enters gain change mode]
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Link to some bench measurements of the PR-2 over at Audio Science Review, done by jerryfreak from a music taping perspective, which usefully includes comparison to Tascam DR2d: https://www.audiosciencereview.com/forum/index.php?threads/deity-pr-2-portable-audio-recorder.56098/ (https://www.audiosciencereview.com/forum/index.php?threads/deity-pr-2-portable-audio-recorder.56098/)
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Found one of my microdot Y adapters and did a quick initial test of DPA 4060 > PR-2 last night. It works. Plugged in headphones and playback directly from the PR-2 sounds good. Playback control is rudimentary, but functional.
Tonight I plan to record some music from the stereo and also while clapping as loudly as possible, then transfer the files to the computer to have a look at the waveforms and check peak levels in comparison to the the levels displayed on the VU meter, to get a better feel for the appropriate gain settings.
Note on the display-
There is a menu switch that turns the LED power and record lights on/off. However, the display remains on and illuminated at all times unless AUTO-LOCK (hold) has been enabled in the menu (set to either 15sec or 1min), in which case the display turns off when the recorder auto-locks. The auto-lock engagement count down timer is the only way to turn off the display. Once off, there is no indication the recorder is powered up or doing anything. When the display is off, a single press of either the record wheel button or the power button immediately turns the display back on, after which it will remain on for either 15sec or 1 min, although the recorder remains locked.
Would be nice if a press of either button would immediately turn the display off again, or if it could be set to remain off until a triple click of the power button unlocks the recorder. Such functionality may not be overly important to stealthers with the recorder hidden in pocket, but it would be good to to be able to immediately black out the display with a single button press after a visual check. Seems to me that enabling a way to avoid the display from turning on with any inadvertent button press would be important for its intended "on talent" when on-camera / on-stage.
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I believe all gain on the PR2 is digital, so best bet imho is to run at 0, then normalize in post.
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I assume so too. I just want to use the digital gain to match the available dynamic range of the recorder (when in stereo 24 bit mode) to the dynamic range of the mics as optimally as possible.
The measurements over at at ASR indicate the PR2 achieves a DR of 92.9dB when gain is set to 0dB. DPA's specified DR for 4060 Core exceeds that at 106dB, but that reflects the difference between A-weighted equivalent self-noise of 23dB (typ.) up to a 129dB peak SPL @ 1% THD. I figure the real world DR of the mics is closer to the difference between the ITU-R BS.468-4 equivalent self-noise of 35 dB (typ) up to the 126dB RMS SPL @ 1% THD, which works out to 91dB. It fits if the gain is set optimally, in which case the mics rather than the recorder should be the bottleneck in the recoding chain.
Maximizing the available DR probably doesn't matter for most of what folks are taping, and won't for these upcoming shows I mention, but will for the classical recording I plan to do, and is good practice in general.
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I also ran it once with a Baby NBox recording close to the source
and think running about +9dB was about right to avoid go over -10dB on the display
I think in MOST situations I would go +12 or even 15dB
Did you run mic or line in? With the babynbox into the A10 I have always used the line in...
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I also ran it once with a Baby NBox recording close to the source
and think running about +9dB was about right to avoid go over -10dB on the display
I think in MOST situations I would go +12 or even 15dB
Did you run mic or line in? With the babynbox into the A10 I have always used the line in...
there is NO WAY I could use Mic In
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there is NO WAY I could use Mic In
Just to make sure i understand you correctly… you mean because the signal from the babynbox would be too hot for nic in, right?
I seem to remember that people said to use mic in back when the babynbox came out, but i always got a hot enough signal to use line in… i almost always tape amplified shows, though…
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considering I'm at 2 or 3 running Line IN...
there is NO WAY I could use Mic In
Just to make sure i understand you correctly… you mean because the signal from the babynbox would be too hot for nic in, right?
I seem to remember that people said to use mic in back when the babynbox came out, but i always got a hot enough signal to use line in… i almost always tape amplified shows, though…
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Haven't had a chance to really listen or examine to the initial concert recording I made last Monday night, but some initial testing beforehand and a brief listen afterward indicates DPA 4060 may be too sensitive for the PR-2 for our uses. DPA 4061 should probably be used in its place. In testing beforehand, with PR-2 input gain set to 0dB, loud clapping immediately next to the mics (accommodation of which is the gain setting method I use for classical music and less amplified performances) produced an asymmetrical, slightly clipped waveform. On a brief headphone listen directly off of the itself recorder after the concert I heard distortion in the bass.
Will give a more serious listen over the weekend and and examine the waveforms. I'll also do the same loud clap test using the same mics through the DPA XLR adapters into a P48 input on another recorder to compare, as I've not actually used this pair of mics previously. This was the first time I've used the CORE amplifier version of the 4060. CORE is spec'd as improving distortion specifications and SPL handling capability over the legacy version of the microphone that I am familiar with, and I know from experience that the loud clap test can be accommodated with legacy 4060 > CA-UGLY > DR2d.
The concert was bass heavy from the recording position in the pit, but wasn't overly loud. I wore earplugs yet most others around me did not. I intended to note SPL levels using a phone app during the concert yet forgot to do so.
Some concerning behavior- During initial testing beforehand I used a pair of well used Eneloop NiMH. For the concert I switched to a pair of Energizer disposable lithium AA's - not sure how old, but they had been used previously in a DR2d. With the menu entry appropriately set to lithium, the PR-2's battery meter indicated full charge beforehand. I had the LEDs on PR-2 turned off, and the LOCK function set to 15sec. As mentioned previously, lock function blacks out the PR-2 completely (in this case after 15sec) until either of the buttons are pressed. I recorded the opener, then started a new file during the intermission between acts. Sometime prior to the main act the PR-2 no longer responded to any button presses. I assumed the batteries had died, and took myself to task a bit for not bringing spare AA's. At several points during the headline performance I tried unlocking it / turning it back on, in the hope that the batteries might have recovered sufficiently to run it for just a few minutes, but saw no indication on the display. Unlike my typical 4-channel stealth method, I was running the mics pseudo-binaurally in glasses, and figuring the recording had stopped, I at some point took the glasses off my head and let them hang freely on my chest. Afterward I installed fresh batteries, restarted the PR-2, took a look at the files and was surprised to find that the recorder had remained running throughout the headline set even though the display remained dark and I could not unlock the PR-2. Behavior with fresh new batteries installed was normal. I've not explored this further yet. Might reflect PR-2's behavior prior to low battery voltage shutdown. I want to test that further, and also re-run the clap test and check for low frequency distortion while using higher nominal voltage alkaline AA.
Just wanted to relay this initial use report to the thread, more on all this once I get a chance to look into it further..
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Just a thought, sometimes a recording will reveal things that exist, but are masked to our ears
Vocal distortion is at top of mind, where bass harmonics and other instruments wash over it
Then the pre-FM or SBD circulates, and it is all to obvious, and not in a taper's control
The microphone is better than the ear, or at least significantly different, for good or ill.
On the Diety ... So a 20db or 10db pad?
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there is NO WAY I could use Mic In
Just to make sure i understand you correctly… you mean because the signal from the babynbox would be too hot for nic in, right?
I seem to remember that people said to use mic in back when the babynbox came out, but i always got a hot enough signal to use line in… i almost always tape amplified shows, though…
Yeah, for amplified shows, use line in. And here's the math to back it up. ;)
Schoeps MK4s have a sensitivity of 15 mV/Pa. Unfortunately, Deity doesn't provide meaningful specs for the inputs. So let's instead see why running line in with a Sony A10 is a good idea! The A10's mic in is rated for 2.5 mV. With a Schoeps mic, that equals 2.5/15 = 0.166 Pa. Now 0.166 Pa is equivalent to 78 dB SPL (here's a great calculator (https://sengpielaudio.com/calculator-soundlevel.htm)!). Most likely, your typical amplified concert is louder than that and your mic inputs will get overloaded. The line in is rated for 2 V. That equals 2/0.015 = 133 Pa, or 136 dB SPL. So the chances of overloading the line inputs are practically zero (at 136 dB SPL, chances are you'll run away from pain anyways).
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there is NO WAY I could use Mic In
Just to make sure i understand you correctly… you mean because the signal from the babynbox would be too hot for nic in, right?
I seem to remember that people said to use mic in back when the babynbox came out, but i always got a hot enough signal to use line in… i almost always tape amplified shows, though…
In the Tascam DR-2D with the babynbox I run mic-in low gain and it's not too hot. I run mostly amplified sources too. Pretty sure the Zoom F-3 it doesn't seem to matter with the 32 bit float.
Back on topic, I am still not convinced this unit is a material upgrade except for size compared to the Tascam. I was hoping it would be.
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In the Tascam DR-2D with the babynbox I run mic-in low gain and it's not too hot. I run mostly amplified sources too. Pretty sure the Zoom F-3 it doesn't seem to matter with the 32 bit float.
Back on topic, I am still not convinced this unit is a material upgrade except for size compared to the Tascam. I was hoping it would be.
Yeah, it really depends on the rating of the recorder's inputs. The DR-2d's mic inputs are rated up to 158 mV, which is equivalent to 114 dB SPL with Schoeps mics. I don't know about the U.S., but amplified shows here in Europe usually run at around 100 dB SPL these days (even though that figure is a-weighted and therefore, real levels could be higher). As I said earlier, the A10's mic input is much worse. And Deity doesn't even specify this (which says a lot).
And I agree that the Deity is not much of an upgrade compared to what we already have. And the fact that Deity doesn't have a meaningful spec sheet doesn't really inspire a lot of confidence in their recorder.
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In the Tascam DR-2D with the babynbox I run mic-in low gain and it's not too hot. I run mostly amplified sources too. Pretty sure the Zoom F-3 it doesn't seem to matter with the 32 bit float.
Back on topic, I am still not convinced this unit is a material upgrade except for size compared to the Tascam. I was hoping it would be.
Yeah, it really depends on the rating of the recorder's inputs. The DR-2d's mic inputs are rated up to 158 mV, which is equivalent to 114 dB SPL with Schoeps mics. I don't know about the U.S., but amplified shows here in Europe usually run at around 100 dB SPL these days (even though that figure is a-weighted and therefore, real levels could be higher). As I said earlier, the A10's mic input is much worse. And Deity doesn't even specify this (which says a lot).
And I agree that the Deity is not much of an upgrade compared to what we already have. And the fact that Deity doesn't have a meaningful spec sheet doesn't really inspire a lot of confidence in their recorder.
I don't know, and you all are welcome to disagree, but I think the Deity is something special. The preamp seems to me to be really crisp and clear. After playing around, I am only running 24 Bit as I want 2 channels. I am running PR-2 >Ca 11. That's it. And at 2 loud shows this week, I had the settings on +15 dB, which is a lot of gain. That is running mic in to get the 5v PIP. The battery life is great, I am on full with rechargable lithiums after 2 full shows. Now the manual tells me very little, and this has all been through trial and error, but I have 2 personal conclusions from my experience.
a. This is one small >:D setup
b. The recordings sound really good.
It sounds like others are not having the same experience, but I am really impressed.
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The primary advantage is the 5v power. If you are using a different way to power your mics there’s no longer a huge benefit.
That said, the kk14s and pr2 fit in a wallet with a drivers license and credit card etc to make it really easy to Jedi mind trick your way in anywhere.
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To run Schoeps I still need the babynbox or equivalent though. So its Deity or DR-2D, Getting the DR-2D in is easy for me. What I was ready to buy was the Zoom F-3 in a Deity for factor. Stereo 32 bit float and modern features. All of this gear meets a need for someone. Just disappointed it doesn't meet mine.
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Daspy, because you still need Nbox to power your mics, I see no real advantage of PR2 for your use .. other than PR2 being less than half the size of DR2d.. and the locking input jack.
In the Tascam DR-2D with the babynbox I run mic-in low gain and it's not too hot. I run mostly amplified sources too. Pretty sure the Zoom F-3 it doesn't seem to matter with the 32 bit float.
The overload is occurring prior to the digital conversion and really has nothing to do with 32bit float on the F3. It's just that F3 and DR2d are capable of accepting higher analog input voltages than PR2.
Along with Sebastian I'm disappointed by the lack of important specs.
The primary advantage is the 5v power. If you are using a different way to power your mics there’s no longer a huge benefit.
^This. To my way of thinking the advantage of PR2 is mostly about its smallness as an "all in one" via elimination of the need for an additional battery box or preamp. Also agreed with Dallman that it is easy to use, works well, and sounds good within its input limits. Unfortunate its limits seem insufficient for the mics I'd prefer to use..
I didn't get a chance to do the further testing I discussed in my previous post this past weekend, but hope to one evening this week. I did confirm that as the batteries were dying, the PR2 continued running for several hours after becoming totally unresponsive, which was somewhat longer than the show, and that the distortion slowly increased over that time period until the PR2 eventually shut itself down (after properly saving the file). At the start of the recording the overload distortion was driven solely by low frequency drum and bass impacts (unfortunately along with higher frequency harmonic artifacts) slowly progressing into to a very scratchy sounding full range distortion later on as available battery voltage/current continued to drop further.
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^ the unresponsiveness of the recorder after a certain point is concerning. Good to know that it completed saving the file before it shut itself down but not being able to get it to wake up so you could change the batteries is not great.
I like new toys and this is ridiculously small but I think I'll stick with the battery box and A10 for now. If I used DPA 4061 I would for sure be getting one of these.
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Yeah, I've not yet confirmed what happens to the file currently being written if you yank the batteries while the recorder is still running but unresponsive, since at that point you're unable to properly shut it down properly.
As mentioned, for use by my newbie recording friend while I'm away I'm probably just going to switch from 4060 to 4061, which should work for the show he is attending. Just need to unrig that 4061 pair from another setup, which is a PITA I was hoping to avoid.
May do some testing using different batteries to see if the different nominal voltage of alkaline vs NiMH vs lithium makes a difference, despite setting the menu appropriate for the battery type being used, which I suspect simply affects the calibration of the battery meter/light. I haven't looked into this yet, but what's up with the Deity AA batteries that are supplied with the unit? There is a special setting just for those. Some alternate type of lithium chemistry it seems. Anyone have insight into those?
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Looks like the Deity AA batteries are a rebrand of the lithium primary cells of the same capacity made by Nice. They have pretty typical specs of most lithium cells (i.e. Energizer) which makes me curious about the special battery setting in the recorder. Maybe these particular cells have a slightly different discharge curve.
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A couple of ebay locking "Y" adapters arrived. They are 2x female microdot > locking TRS mini plug for ~$11 ea.
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^ Incredible deal. Way cheaper than cutting up those $50 pre-made cables.
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^ Incredible deal. Way cheaper than cutting up those $50 pre-made cables.
But the longer microdot connectors, instead of the shorter ones more like dpa and cdi use.
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Connector is about the same length as the 18 year old DPA microdot extension cables I have on hand, which are male on each end and use barrel connectors to mate with the male microdot on the mic cable. See attached photo comparing the DPA extenstion cable (with barrel installed), next to the Ebay Y adapter (the blue jacketed cable). I can similarly compare against the CDI Int cables, but as I recall the connector lengths on them are similar. The main differences are that the CDI cables are considerablay more flexible than the DPA extension cables and can be orded with female microdots, eliminating the barrel connector.
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This is the modern cdi/dpa female end I am way more comfortable with.
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The connectors on your pair with the red heat-shrink look identical to the connectors on the Ebay Y to me.
FYI- In your photo, I consider those to be the female end (as does CDI Int when ordering cables from them), which references the the gender of the center conductor, rather than the outside threaded part (yeah, its confusing).
The locking Ebay Y's worked well in the two test outing so far.
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Yeah the photo was comparing the eBay ends to the cdi ends.
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I'll have to check my CDI ends, but they look different than yours.. doesn't have heat shrink wrapping around to the threads.
I ran a second test of PR2 with DPA omnis a free outdoor amphitheater show last Saturday. Had my open rig rolling as well, so was able to play around with the PR-2 in the head-worn rig, swapping between DPA 4061 and 4060-CORE, and swapping between the Deity batteries and Energizer Lithiums (a fresh, unused pair this time), as well as moving between my open recording position, center of pit, and directly in front of a PA stack in search of highest SPL. PR2 input gain remained set to 0dB the entire time.
The 4061's worked fine, with an output level which seems appropriately matched to the input stage of the PR2.
Switching to 4060_CORE produced some audible bass distortion again, particularly when standing directly in front of the PA/subwoofer-stack in a typical close stack-taping position. Using new batteries, the distortion at this show was significantly less bad, yet nether set of fresh lithium batteries completely eliminated the overload problem as hoped. Perceptually this concert was significantly louder than the previous test I did several weeks back at the larger amphitheater event with what turned out to be dying Energizer lithium AAs, certainly so when standing adjacent to the stack. How loud was it? Not sure. I used a sound meter app which hovered around 91dB while standing at the stack, but it registered approximately 90dB back at my open recording spot a few rows behind the pit and that doesn't add up. Doesn't add up as obviously it was significantly louder directly in front of the PA. SPL measurement was taken using a new to me Galaxy S24.. and I don't trust the SPL app.. I see no way of determining what filtering it is using, and it seems it might have been be hitting some kind of hard limit possibly imposed by the phone. Dunno.
Have yet to measure the mic power voltage out of this particular PR2.
Prior I did do a quick test comparing both sets of mics plugged directly into PR2 (at 0db input gain) verses recording via DAD6001 phantom adapters into R-44_OCM. I used the closest equivalent input gain setting available on the R44, which was -2dB. There was significantly increased headroom using the DAD6001 adapters powered with P48, but I've not yet quantified how much.
I now feel I've at least done enough testing to confirm use of 4061 > PR2 will be the best option for use by my newbie recording friend while I'm away. I'll report on how well that worked out, and continue with further testing after I return mid October.
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Here are a few photos of the rig configured for use by my non-taper friend. I used some "soft, quiet" techflex intended for body harnesses that is less plasticy and more of a fabric weave to manage the cables, streamline things, and make use a bit more comfortable .
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Hear are a few photos of the rig configured for use by my non-taper friend. I used some "soft, quiet" techflex intended for body harnesses that is less plasticy and more of a fabric weave to manage the cables, streamline things, and make use a bit more comfortable .
Nicely done!
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Does the PR-2 have a Bluetooth app or not? There's no mention of an app in the owner's manual?
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Does the PR-2 have a Bluetooth app or not? There's no mention of an app in the owner's manual?
The Sidus Audio app is used by the PR-2 and in my experience it works very nicely with my android phone.
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Does the PR-2 have a Bluetooth app or not? There's no mention of an app in the owner's manual?
The Sidus Audio app is used by the PR-2 and in my experience it works very nicely with my android phone.
I'm not familiar with that app. Is it difficult to set up and use?
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I've not tried the Sidus Audio app yet, but I intend to. I'll also be using it on Android. That will have to wait until my return mid Oct.
Dalman- I take it you've not had any obvious problem with dropped samples as was reported over at Audio Science Review when using the Sidus Audio app.. I don't know the details, might have been doing so when the bluetooth connection is lost and it reconnects again. Dunno. I'll test that at some point, but am curious to hear more about your experience thus far.
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Follow up question: It appears from the specs that the 32-bit float is available only in mono, correct? For 32-bit float stereo, there's just the Zoom H1 Essential, and soon the new Tascam?
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Follow up question: It appears from the specs that the 32-bit float is available only in mono, correct? For 32-bit float stereo, there's just the Zoom H1 Essential, and soon the new Tascam?
no 32bit stereo on the pr2
32bit stereo on the sound devices mixpre ii series and the zoom f series
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Yep. PR2 produces a 24bit file when set to record two channels / stereo (even though it really only provides performance specs equivalent to around 16bits)
For 32-bit float stereo, there's just the Zoom H1 Essential, and soon the new Tascam?
As far as the category of small, 2ch only, 32bit-fp recorders goes, there are currently a few different models offered from Zoom I believe. The Zoom F3 was one of the earliest and I believe still has the best specs.. other than MixPre3ii which is larger (especially compared to PR2), which is the smallest recorder SoundDevices offers currently. We are very likely to see more small recorders show up in this category.
Edit to add- Will be interesting to see how good the new Tascam is, but we know it will feature XLR inputs just like F3 and H1E so no question it is going to be significantly larger than PR2. Hopefully Tascam is able to produce a small 32bit float recorder with input range specs that are at least equivalent to the old non-XLR input DR2d, first produced 14 years ago. Sad that newer manufacturers don't seem able to match much less surpass that in a similarly priced recorder today.
PR-2 is somewhat unique in its combination of featuring minijack input only (making it much smaller than it could otherwise be if it had XLR inputs) and providing 5V mic power. The attraction for most folks following this thread is the combination of those two features.
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IMO, 32 bit float isn't that essential for these small recorders, but a control app becomes important.
Running my mixpre6, I don't run too hot, don't need the ability to over-record.
Sony a10 w mics chopped is quite small and bluetooth app is rock solid.
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Follow up question: It appears from the specs that the 32-bit float is available only in mono, correct? For 32-bit float stereo, there's just the Zoom H1 Essential, and soon the new Tascam?
Also Zoom F3
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Yep. PR2 produces a 24bit file when set to record two channels / stereo (even though it really only provides performance specs equivalent to around 16bits)
Hey Gut! I appreciate all the testing and reporting you and everyone else has done on the PR-2. But I was a bit confused by your statement about performance - I've read the thread but I don't quite see what you mean by "equivalent to around 16 bits." Are you specifically referring to recording low volume performances? or the overloading you saw with DPA 4060s?
And grawk said "I believe all gain on the PR2 is digital, so best bet imho is to run at 0, then normalize in post." Is that your recommendation as well?
I'm considering a PR-2 to replace my antique 16 bit iriver H320 and running my CA-11 mics straight in without Chris' preamp/battery box. Others have said this setup works well for loud shows.
Thanks!
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I've not tried the Sidus Audio app yet, but I intend to. I'll also be using it on Android. That will have to wait until my return mid Oct.
Dalman- I take it you've not had any obvious problem with dropped samples as was reported over at Audio Science Review when using the Sidus Audio app.. I don't know the details, might have been doing so when the bluetooth connection is lost and it reconnects again. Dunno. I'll test that at some point, but am curious to hear more about your experience thus far.
Easy to use and robust. I have not had issues with it al all.
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Yep. PR2 produces a 24bit file when set to record two channels / stereo (even though it really only provides performance specs equivalent to around 16bits)
Hey Gut! I appreciate all the testing and reporting you and everyone else has done on the PR-2. But I was a bit confused by your statement about performance - I've read the thread but I don't quite see what you mean by "equivalent to around 16 bits." Are you specifically referring to recording low volume performances? or the overloading you saw with DPA 4060s?
Reflects its actual, real-world dynamic range capability (from noise-floor to max level before clipping) when recording in stereo mode and producing a 24bit file output. See the bench testing done over at Audio Science review for details. A link to that can be found in one of my earlier posts in this thread.
And grawk said "I believe all gain on the PR2 is digital, so best bet imho is to run at 0, then normalize in post." Is that your recommendation as well?
Yes as long as the output level of the microphone does not overload the PR2 when it is set to 0db gain. That's the problem I was having with DPA 4060 which has a sensitivity of 20mV/Pa. Setting PR2 gain lower than 0dB gain in an attempt to accommodate that higher output level from the mic was reported to not increase the PR2's input level capability any further. My testing seemed to confirm that but need to double check to make certain.
You should be able to use some positive input gain as necessary when using microphones that have a lower sensitivity and thus a lower output level, in order to achieve optimal gain-staging and make the most of the available dynamic range. Doing that should provide better low-level noise performance verses normalizing an overly low recorded level afterward.
I'm considering a PR-2 to replace my antique 16 bit iriver H320 and running my CA-11 mics straight in without Chris' preamp/battery box. Others have said this setup works well for loud shows.
So far it seems okay with DPA 4061, which has a sensitivity of 6 mV/Pa. If your mics have a sensitivity of around that or less, and operate correctly when supplied with 5V, I suspect you should be good with PR2
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I've not tried the Sidus Audio app yet [..snip..] curious to hear more about your experience thus far.
Easy to use and robust. I have not had issues with it al all.
Good to hear. I look forward to trying it in October.
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I’m seriously considering impulse buying one of these after reading through this thread. I’ve been wanting a very “easy” stealth rig for when I’m traveling, and this seems like it would be a great solution. I don’t like having to keep track of my gear when I’m on a trip that might be a couple weeks long, so a small recorder of this size plus a pair of mics (And nothing else) really appeals to me. I’m heading to Japan next week and that would be a great opportunity to test this out if I do.
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I’m seriously considering impulse buying one of these after reading through this thread. I’ve been wanting a very “easy” stealth rig for when I’m traveling, and this seems like it would be a great solution. I don’t like having to keep track of my gear when I’m on a trip that might be a couple weeks long, so a small recorder of this size plus a pair of mics (And nothing else) really appeals to me. I’m heading to Japan next week and that would be a great opportunity to test this out if I do.
This is why I purchased the deck, and I have been very pleased.
Here is a sample:
https://archive.org/details/maggierose2024-08-25.Deity (https://archive.org/details/maggierose2024-08-25.Deity)
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I’m seriously considering impulse buying one of these after reading through this thread. I’ve been wanting a very “easy” stealth rig for when I’m traveling, and this seems like it would be a great solution. I don’t like having to keep track of my gear when I’m on a trip that might be a couple weeks long, so a small recorder of this size plus a pair of mics (And nothing else) really appeals to me. I’m heading to Japan next week and that would be a great opportunity to test this out if I do.
This is why I purchased the deck, and I have been very pleased.
Here is a sample:
https://archive.org/details/maggierose2024-08-25.Deity (https://archive.org/details/maggierose2024-08-25.Deity)
Samples sound great - thanks for sharing! That’s almost the setup I’ll be running here, CA14 cards into the Deity. I did order it and out for delivery today, perfect timing as I leave for Japan on Tuesday. Looking forward to testing this thing out.
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Reporting back after my first test of this thing…I’m happy with it in short. Easy to use once you know the controls, and the iOS app is pretty intuitive other than the sliders for mic gain. It was a heavy metal band I taped in an arena, 30 feet or so from the stacks. There was no support act to gauge my levels, so I went in as blind as the possible. I did some screwing around with my gain during the intro tape trying to find a sweet spot and settled on +12 in each channel, that still only yielded peaks around -15 db I think. The only things I’m struggling with on the device are playback directly from the device, and transferring to a computer via USB-C (I opted to use a microSD adapter to transfer the file). Spot checking the recording I’m quite happy with it given the real bare bones rig of CA-14 cards right into the Deity. I’ll post a sample when I get a chance if there’s any interest. May be a few days, as I have 8 events in 8 days so quite a busy schedule while I’m abroad.
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A sample of my first test on the Deity. Recorded about 30 feet back and slightly inside from the stacks - unfortunately in Japan you don't necessarily get the luxury of picking a prime taping spot. This is just the raw file - I haven't amplified it, done any noise removal, etc. - exactly what I captured.
https://mega.nz/file/QF8BHJIC#nmm9HcXwT5Cy57y_2bFDapWv4Up8PcjQko_Lx25gouY
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I also got a chance to try out my new Deity PR-2 last weekend.
I recorded a couple of shows and the unit seemed to perform flawlessly. I'm also running Church Audio CA-11 mics straight into the PR-2. I confirmed that the PIP is 5V (and not adjustable as far as I can tell...)
The unit is very small. I brought it into the United Center in Chicago in a small bag with some other stuff and no issues.
Here are some tidbits:
edit - APP version 1.6, HW version 1.0
1. As noted earlier, the battery compartment door is very tight. So tight that I'm afraid to completely close it - I leave enough of a gap that I can get my fingernails in to wrench it back open. The batteries are also extremely tight and take a lot of force to get them in. I've tried NiMH and standard alkalines. Both very tight. Have not tried the supplied Deity lithium batteries...
2. The unit automatically splits long recordings into separate files. The split point is just past 4 hours (4:08:32 in my test). The file size was 4,194,001 kb. The rejoined files sounded smooth across the split point.
3. If you pop a battery out during a recording, the unit saves the file and then shuts off. I shoved the battery back in after 5 seconds, but the unit was already off.
4. The Deity runs its date clock even without batteries for at least a few days. I doubt it would keep the date function current forever, but you certainly won't lose the date if you swap batteries. The unit saves files in folders for each date as "PR-2 -001"; "-002", etc. In each folder, the file name sequence starts over. You've got to be a bit careful when you transfer several "PR-2 -001" files recorded on several different dates.
5. The gain controls for L and R do not link. If you're recording and need to change gain, the changes will occur on each channel, independently, as you spin the gain dial.
6. The recordings I made sound fine, and I was nowhere near clipping the analog inputs of the Deity. But, I have no idea where the CA-11 mics will start to clip with 5 volts. I've read that they have less than 0.6% THD at 114 dB at 1000 Hz with 9 volts. Not sure how to get to those volumes with a home system...
7. I recorded the shows at the United Center at +12 Gain. According to the Audacity "Amplify" effect, I could have safely run at +18 Gain with 6 dB to spare.
Now I just need to get a locking 3.5mm plug and I'm all set!
edit - Locking 3.5 mm plugs that "look" like the same one that's on the included lav mic:
https://www.amazon.com/Connector-Locking-Microphone-Adapter-elPart2334555/dp/B0CRQXR7ZP
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A sample of my first test on the Deity. Recorded about 30 feet back and slightly inside from the stacks - unfortunately in Japan you don't necessarily get the luxury of picking a prime taping spot. This is just the raw file - I haven't amplified it, done any noise removal, etc. - exactly what I captured.
https://mega.nz/file/QF8BHJIC#nmm9HcXwT5Cy57y_2bFDapWv4Up8PcjQko_Lx25gouY
Thanks. Do you recall where your gain was set? This clip peaks at -16. What mics? I recently picked one up and have yet to run for a concert. I use modded AT853's and usually a Roland R-07. I am able to fit my mics and the PR-2 in a Kangol. That was my goal.
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A sample of my first test on the Deity. Recorded about 30 feet back and slightly inside from the stacks - unfortunately in Japan you don't necessarily get the luxury of picking a prime taping spot. This is just the raw file - I haven't amplified it, done any noise removal, etc. - exactly what I captured.
https://mega.nz/file/QF8BHJIC#nmm9HcXwT5Cy57y_2bFDapWv4Up8PcjQko_Lx25gouY
Thanks. Do you recall where your gain was set? This clip peaks at -16. What mics? I recently picked one up and have yet to run for a concert. I use modded AT853's and usually a Roland R-07. I am able to fit my mics and the PR-2 in a Kangol. That was my goal.
Had that info on the previous page, but to reiterate it here I was running CA-14 cards direct into the Deity. I had the gain set to +12. Not sure if you've played around with the unit at all, but a head's up also that the gain can only be set in 3 dB increments.
I've since taped two more shows with this rig - I taped another in Japan, through the barrier directly in front of the PA and pulled a great recording. I also used it for a little dive bar gig the other night also, haven't had a chance to check that recording yet. But I love how easy this thing is to use, and been quite happy with my results so far.
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I've since taped two more shows with this rig - I taped another in Japan, through the barrier directly in front of the PA and pulled a great recording. I also used it for a little dive bar gig the other night also, haven't had a chance to check that recording yet. But I love how easy this thing is to use, and been quite happy with my results so far.
Right on! Thanks for the insight. I just got in from John Scofield. I was able to run open from my seat, so I ran km184 > F8. I also put out my AT835 > Deity PR-2 as a test. I put the recorder at +18 and ended up with peaks at -7. This was a club gig with pretty low SPL's. Came out pretty nice actually. I do wish it could do 24/96 though. Considering I will use it as a LAV for video work, it was a great purchase. I have a second one coming. Might have to try dual 32b recording.
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I have a second one coming. Might have to try dual 32b recording.
It will be very interesting to hear your experiences.
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Well, upon my return to the US a couple weeks back, I was really hoping to enjoy a recording of BEAT made from a primo recording seat, then take a deeper look and discuss how well the PR-2 rig pictured a few pages back worked in that situation. Indeed, that show was my primary motivation to setup the DPA4061>PR-2 rig in the first place. Unfortunately my highly motivated friend ended up unable to attend, but went to admirable lengths to pass the rig and the very simple instructions for its use (put glasses on head, turn on recorder, put recorder in pocket) off to anther friend who was able to attend.. but did not bother to take the rig with him. Harrumph.. and c'est la vie.
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You mean you have to take this rig with you in order to get a recording? What sort of punk old-school tech is this??
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You must be present to win!
Well, I decided a recording of that concert and of others to follow was the primary goal of this thing, so decided not to take the rig along with me to record city, village, mountain and jungle ambiences and any music performances I happened to come across. I did briefly consider picking up a second PR-2 to put together a rig using the more sensitive 4060CORE pair to take with me, but decided my new phone would cover things well enough. Now wish I had done that as I would have had it rolling when in the forest of Chitwan when an elephant walked right up and started purring and talking, and especially a bit later when a deer started wailing some funky warning cries followed by the roar of a tiger nearby in the bush. An "all hairs up on the back of your neck" deep sound! Phone video caught it, but is a static shot of the bush, with the roar audible if listening closely about 20 sec in. Cool, but an "as if you were there" pseudo-binaural recording would've conveyed the live experience if it far better. On the way back out of the forest we came across some impressively large tiger paw prints which weren't there on the way in, but never did get a visual on the big cat.
Meanwhile on this side of the planet, Adrian Belew was effecting some Elephant Talk himself, and while he didn't perform Big Electric Cat, he and those other cats certainly roared impressively! ..at least so I'm told.
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I guess I should share my experience. I mentioned above that I would be getting a second one. The first one I bought was from B&H and the shipping and retail packaging was fine. The recorder had a small break/crack at the seam. IMO would have been packaged that way. It's cheap plastic and Chinese manufacturing. I had written the company and sent an image. They asked for a video clip, which is fine. They then asked me to take it up with B&H. B&H wanted pics of the shipping package and retail package. I didn't have the shipping package by the time this came about. I went back to Deity and gave them the update. Letting them know that neither packaging was damaged and it was packaged with the flaw. They did the right thing and offered to ship a new unit. It was shipped out from China and arrived today. First tiny miff (and this is not a big deal) is that they did opt to not include the batteries or the lav. I get that but all things considered it was a small bummer. I guess that's selfish of me. The second small grip is that this new one, I can really hardly seat the battery. I first tried a Duracell and then went to the Deity batts from the first unit. All batts go in and out of the first unit fine. It's a massive struggle to get them into the replacement unit. It seems like the coil is a slightly higher gauge metal on the replacement. Then, once I finally cram it in, the cover doesn't want to slip in. With batts in it's at least twice as hard to open and close as the original unit. Oh well... Just sharing/venting.
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it seems reasonable to me that they just replaced the broken part, and not the accessories, assuming you weren't asked to send anything back.
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Just received mine. The battery cover is a nightmare.
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I will echo the sentiments that the battery cover is a really tight fit. Mine has gotten a little easier to use after being slid on/off half a dozen times, but it’s still tighter than would be ideal.
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Mine also tight, but not frustratingly so. Optimistically, I figure that's not a bad thing as its likely to wear in with some use until it fits just about right. I've plenty of similar devices where the removable battery cover has become overly loose after some of use and requires gaffer tape to hold it in place.
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it seems reasonable to me that they just replaced the broken part, and not the accessories, assuming you weren't asked to send anything back.
Yes, of course. I probably should have just kept my mouth shut on that.
On a related note. Does anyone want to part with the LAV that comes with this recorder?
edit: bought Grawks lav
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I’d sell mine. I just have to find it
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I got my hands on a PR-2 today. Here are my observations:
- Overall the unit looks and feels better than expected. Even the battery cover is not that bad on my unit.
- My Ni-MH rechargeables are a real tight fit, but I can still manage that.
- I can't get the Micro-SD card in and out of the slot without a pair of tweezers. I don't like that.
- Battery run time is fantastic!
- The Bluetooth connectivity feels solid. I can re-start the iOS app after some time and it is still connected. This means I can start recording right away. This is as good as the Sony app. And this is where Olympus dropped the ball with their app.
- The app feels solid. The only thing that's exremely stupid is that gain for the left and right channels can't be linked. Also, the level meters only go down to -40 dBFS. I would have preferred -60 dBFS.
- I love the locking 3.5mm inputs. However, non-locking Neutrik right-angle plugs do tend to slip out of the jack easily.
- The PR-2 feels very small, but I do prefer the Sony A10 form factor.
- I don't like the fact that the unit is 48 kHz only. I would have preferred to be able to set it to 44.1 kHz as that is all I've used for years.
I also did a few quick measurements:
- The gain settings are accurate. At the 0 dB gain setting, a 0 dBU (775 mVrms) input signal results in a recording with a 0 dBFS peak. A -12 dBU input signal results in a recording with a -12 dBFS peak and so on.
- Equivalent input noise (EIN) is -98 dBU (unweighted; 150Ω source resistance; +36 dB gain; line in) and -106 dBU (A-weighted; 150Ω source resistance; + 36 dB gain; line in). Deity's spec'd -130 dBU EIN looked suspicious from the get-go. The numbers I measured look more like what I'd expect from a recorder of this class. For our purposes, this is probably not a problem in any real-world scenario. It's just a bit odd that they would list a EIN value that even way more professional and way more expensive gear can't achieve.
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I thought maybe, just maybe, the "pass through" of power might enable support for mics that require/prefer higher than 5V PIP. It does NOT. Here's a snippet of my conversation with Deity via YouTube comments.
My Question:
Question... Your site makes the following statement "The PR-2 features a toggle switch on the top that allow you to pass the microphone PiP from the output thru to the input. This means you can daisy chain the PR-2 with a transmitter so you can add recording functionality to any transmitter you might own. This also means even if the PR-2 is turned off, the transmitters PiP is still passing thru to power your lavalier."
Can I use this feature to pass slightly higher voltages through to my mic (like 9V)? I record loud sounds which and some mics prefer slighter higher voltages (more than 5v) to avoid distortion. Wondering if passing through 7-9v from the output port might allow me to use the PR-2 and these hungrier mics. Thoughts? Thanks.
Diety Response:
On US units, the pass-through is disabled during recording. 5v is the max it can handle and will NOT support 9v mics.
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what would be the use case for pass through pip in a concert recording situation where you aren't using a wireless transmitter?
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I can envision two scenarios-
Maybe as a cool alternate way to connect a 9V battery box which would eliminate any additional connections in the signal path between mics and recorder. The battery box wouldn't need anything like a transmitter or second recorder attached down stream of it.
OR
Potentially passing through the higher PIP voltage through from a secondary recorder?.. but I'm am unaware of any recorders that produce more than 5V PIP natively, so that higher voltage would need to be supplied from an XLR adapter / PFA in the second recorder.
However, even with a non-US version of the PR-2 those scenarios are not assured to work, as the pass-thru path might be capacitor coupled, just like it is through a battery box, in which case it will pass signal but not DC voltage. Deity's answer didn't indicate if it's actually possible with a non-US version or not, only that it isn't with the US version.
Blakenan, even though you can't plug the battery box into the PR-2's headphone output and have its voltage passed through to the mics, you can still use your 9V battery box in the typical way between the mics and the PR-2.
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I recorded W.A.S.P. and Armored Saint last night using the PR-2. For mics, I used the Sound Pros CMC-8 with the low sensitivity mod and their key fob battery box. The settings on the PR-2 were line in, 75 hz low cut, and +12dB gain on each channel. I was positioned about 20 feet in front of the right speaker stack. When the recordings were put through Audacity, it indicated a volume increase of about +6dB was possible before clipping, so I think I could've set the recorder at +15dB gain without any trouble. I liked being able to monitor the recording through the app rather than taking the recorder out my pocket and actually looking at it, and it was easily concealable and did not set off the metal detectors. I have a doom metal show coming up next week and I might try recording at least one the bands using the PR-2's pip to see how how that turns out.
The W.A.S.P. recording is posted in Kickdown Central if anyone wants to check it out.
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I just posted a benefit concert I recorded with my pr2 and 4015s
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Do we have much insight yet as to the PR-2 and security screening? I have Gilmour in a week at MSG. I might end up taking the train in and realizing this might get complicated. Any insight?
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It’s the easiest recorder I’ve ever tried to get past security. Do what you normally do with gear to get it in.
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Okay, I'll see what I can do. I've made countless hundreds of recordings but 99.99% with permission and whatnot. Very little in this manor, just put together a new rig. I guess I'll just see what happens :)
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I had a PR-2 and my mics concealed on my person at a show I went to last week, and it didn't set off the metal detectors.
Do we have much insight yet as to the PR-2 and security screening? I have Gilmour in a week at MSG. I might end up taking the train in and realizing this might get complicated. Any insight?
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I’ve had what I assume were the batteries set off one metal detector last weekend, but I had it concealed well enough I was able to come up with a good cover, and no further questions asked. When I used it this weekend I took the batteries out and had them in my camera case, and it did not set off a metal detector without the batteries inserted.
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I had a PR-2 and my mics concealed on my person at a show I went to last week, and it didn't set off the metal detectors.
Do we have much insight yet as to the PR-2 and security screening? I have Gilmour in a week at MSG. I might end up taking the train in and realizing this might get complicated. Any insight?
Thanks for chiming in. This is reassuring.
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Do we have much insight yet as to the PR-2 and security screening? I have Gilmour in a week at MSG. I might end up taking the train in and realizing this might get complicated. Any insight?
Just walked a Tascam DR-2D babynbox mk4s and actives in. No issues. Plug a pair of ear buds in it and it's an mp3 player.
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^ Yes.
Anyone have any insight into whether disposable lithium batteries will pass metal detection more easily than alkaline or NiMH or not?
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^ Yes.
Anyone have any insight into whether disposable lithium batteries will pass metal detection more easily than alkaline or NiMH or not?
I think it's the metal shell of the battery that is seen.
Lighter and slightly less dense, lithium may have an advantage.
Metal detectors at looking for weapons above all, but also abnormalities.
If you can just put it in the tray and let them eyeball it, that seems the safest overall.
Recent reports on stage-crashing (Morresey <sp?>) and the slaying of a young performer at her meet/greet signature session might ramp security up everywhere.
Fans are strange.
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^ Yes.
Anyone have any insight into whether disposable lithium batteries will pass metal detection more easily than alkaline or NiMH or not?
Really wish we had someone on this forum with open access to metal detection, who could just do a bunch of test cases for various batteries, rigs etc. haha
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^ Yes.
Anyone have any insight into whether disposable lithium batteries will pass metal detection more easily than alkaline or NiMH or not?
Really wish we had someone on this forum with open access to metal detection, who could just do a bunch of test cases for various batteries, rigs etc. haha
It also largely depends on how sensitive they set the equipment. They can't set it too sensitive, as it would result in too many false positives on every button, zipper, brah. So if recorder 'X' passes in one venue, it may not pass in another venue, even if they use the exact same equipment.
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Really wish we had someone on this forum with open access to metal detection, who could just do a bunch of test cases for various batteries, rigs etc. haha
Have you got one of those pipe/cable detectors in your tool box?
I have and I’ve tested all my recorders and microphones in my pocket and they all beep.
What we can never know is what sensitivity they set their detectors to. Fortunately 90%+ of the gigs I go to in the UK don’t use metal detectors.
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This recorder looks a lot like a guitar tuner....which isn't forbidden in those venues !
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I've just posted my Gilby Clarke Deity PR-2 recording in Kickdown Central if any one wants to hear what it sounds like.
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I've just posted my Gilby Clarke Deity PR-2 recording in Kickdown Central if any one wants to hear what it sounds like.
I'll have to grab this when I get time later next week. Wondering how you felt about the recorder? Levels easy enough to set? I've read that you can't gang left and right (might put in a future firmware request because that should be easy enough.
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I've just posted my Gilby Clarke Deity PR-2 recording in Kickdown Central if any one wants to hear what it sounds like.
I'll have to grab this when I get time later next week. Wondering how you felt about the recorder? Levels easy enough to set? I've read that you can't gang left and right (might put in a future firmware request because that should be easy enough.
Certainly going to be nice and easy to get into venues, although on this occasion, there was no physical search. Level setting nice and easy via the app, it fact, it was all straightforward using the app. I had them set to +15, but could definitely go higher. I'll probably try +18 next time.
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For what it's worth. I went to a local college arena for a concert last night. They tout this on the info page.
New and Improved Patron Screening
The Mullins Center is excited to introduce upgraded weapons detection technology that provides for faster and more convenient security screening for all attendees by not requiring fans to stop at a checkpoint.
Unless instructed otherwise by security staff, guests can keep small personal items (such as cell phones, keys, and wallets) on them while entering.
Guests might be directed to secondary screening, including hand-held magnetometers or visual inspection. Additionally, all permitted bags are subject to inspection.
We ask that you cooperate fully with staff and arrive at the entrances with adequate time for entry.
The line moved fast! I had at853 and PR-2 in pocket and got through no issue.
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Is this really works well with AT853s and 3-wire battery boxes via line-ine mode? The small size is really seducing, I might want to upgrade my stealth rig with this recorder next year.
Great feature it's using AA batteries and not a built in one. Alsos 24/48 is totally okay for me, since I record audio for video.
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Yes. For 3-wire you'll still need the battery box. 2-wire modded can go straight in, powered by the recorder.
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Has anyone tried synchronizing timecode across multipe PR-2 units? In other words: I'd like to find out if it's possible to sync multiple units via timecode so that their clocks stay in sync for 4-channel (2 units) and 6-channel (3 units) recordings. I'm currently using the old Tascam DR-2d for low-profile multichannel recordings, but I'd prefer to use something that can be controlled via Bluetooth.
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Not yet, but that's also my interest.
A reach perhaps, but hoping it may be possible to clock sync two PR-2 via a simple USB connection between the them, without the need for introducing an additional timecode device.
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I'd like to find out if it's possible to sync multiple units via timecode so that their clocks stay in sync for 4-channel (2 units) and 6-channel (3 units) recordings.
Timecode is just metadata (that's an over simplification but close enough). It doesn't actually sync the clocks it just tells you "when" based on a running timeline.
You need wordclock sync to actually sync the clocks to the sample level if that is what you are trying to accomplish.
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The underlying question is- what level of sync is good enough?
In some modes, Timecode can sync clocks to frame-rate level (somewhere around 30 times per second), but not to sample-rate level (which at 48,000 times per second is a couple orders of magnitude more precise). Agreed that a wordclock / sample-level degree of sync is ideally what we'd like to achieve, and is arguably what's necessary across the various microphones of a phase-correlated multichannel stereo array. But how close do we really need and can we achieve that with timecode?
First off, we can allow for somewhat less sync tolerance between sources that are not highly phase-correlated, such as AUD mics + SBD feed, or a pair or room mics that placed separately from the main pair. Secondly, clock chips these days tend to be much more accurate than they used to be, running closer together and drifting less. In the timecode mode where sync is confirmed every 30th of a second or so, that's probably more than sufficient. And in the timecode mode where sync is only jam sync'd at the start, after which each clock runs freely, identical modern clock chips in identical modern recorders are more likely than they used to be to run close enough over the course of one live performance set.
For those reasons I suspect a timecode frame-level of sync will probably be sufficient for most taper use. And an initial clock jam may be enough. But we need to try it to really see.
(https://collection.nam.ac.uk/images/960/79000-79999/79123.jpg)
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I made a couple of recordings with Church CA-11 mics directly into the PR-2 recorder with 5 volts PIP on an outing to Chicago this summer. I think the Church mics are OK with 5 volts, but I can't convince myself that they're as clean sounding as with Chris Church's preamp and 9 volts. But the peaks in these shows were 9-12 dB below clipping and maybe the PR-2 preamp noise crept in when boosted in post.
The Church CA-11 mics are spec'ed to have "distortion less than 0.6% THD @ 114 dB @ 1 kHz". I'm wondering how they'd do with only 5V? Is there an easy way to make that measurement?
Anyway, here are the shows if you're interested. Each has been heavily EQ'ed (for better or worse...)
The National 2024-09-24 United Center, Chicago, IL
Live Music Archive: https://archive.org/details/TheNational2024-09-24
The War On Drugs 2024-09-24 United Center, Chicago, IL
Live Music Archive: https://archive.org/details/twod2024-09-24
Jack Broadbent 2024-09-22 FitzGerald’s, Berwyn, IL
DIME: http://www.dimeadozen.org/torrents-details.php?id=783672
MP3 (127 MB): https://drive.google.com/file/d/1Qsg8Z24dTmT0OGlG4NPmAXX0epsdsXNF/view?usp=drive_link
FLAC (942 MB): https://drive.google.com/file/d/1BldRSd66q8uED2f8RZlZ83_5tOttfRyW/view?usp=drive_link
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The Church CA-11 mics are spec'ed to have "distortion less than 0.6% THD @ 114 dB @ 1 kHz". I'm wondering how they'd do with only 5V? Is there an easy way to make that measurement?
That depends on your definition of "easy".
Theoretically you'll need a signal generator as well as a spectrum analyzer or an oscilloscope with FFT capability. But you'll only be able to measure rather high THD values (> 0.5 %) on the latter.
https://youtu.be/s_cVP5gu4SY
Another way is using your computer and software:
https://www.diyaudio.com/community/threads/how-to-distortion-measurements-with-rew.338511/
There's also specialized measuring equipment, but it's very expensive.
https://www.tek.com/en/products/keithley/digital-multimeter/2015-series
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Thanks for the info. I've got REW and a calibrated mic, so that's part of the equation. I also can get my hands on a pro EV horn and power amp to get safely and cleanly above 110 dB at 1 kHz. I just need to do some research on how to get the signal from the Church mics into REW through my sound card if that's possible. And then figure out what REW is saying about the signal...
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Thanks for the info. I've got REW and a calibrated mic, so that's part of the equation. I also can get my hands on a pro EV horn and power amp to get safely and cleanly above 110 dB at 1 kHz. I just need to do some research on how to get the signal from the Church mics into REW through my sound card if that's possible. And then figure out what REW is saying about the signal...
Just be careful. Pure sine waves are not really pleasant to listen to at room volume, let alone at 110 dB SPL.
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Hi, can someone who owns this device clarify for me please.
If I have a standard set of mics, such as AT831 terminating in a single 3.5mm plug, does the device record in 32 bit stereo, or will it just record one channel mono?
The online specs seem to indicate the later but wanted to be sure.
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Hi, can someone who owns this device clarify for me please.
If I have a standard set of mics, such as AT831 terminating in a single 3.5mm plug, does the device record in 32 bit stereo, or will it just record one channel mono?
The online specs seem to indicate the later but wanted to be sure.
32 bit float is mono
24 bit is stereo
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Hi, can someone who owns this device clarify for me please.
If I have a standard set of mics, such as AT831 terminating in a single 3.5mm plug, does the device record in 32 bit stereo, or will it just record one channel mono?
The online specs seem to indicate the later but wanted to be sure.
32 bit float is mono
24 bit is stereo
Right
So it is taking input from both channels/microphones and mixing down to a mono source? or is it only recording from one microphone and not the other?
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Hi, can someone who owns this device clarify for me please.
If I have a standard set of mics, such as AT831 terminating in a single 3.5mm plug, does the device record in 32 bit stereo, or will it just record one channel mono?
The online specs seem to indicate the later but wanted to be sure.
32 bit float is mono
24 bit is stereo
Right
So it is taking input from both channels/microphones and mixing down to a mono source? or is it only recording from one microphone and not the other?
It records from one microphone. This setting is intended to be used with a single lavalier, but it works with the left mic out of a stereo pair because the signal comes from the tip of the minijack pin.
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Finally unboxed my PR-2 and planning to try out tonight at a rock show. Have been doing some testing at home and have some questions.
DPA 4061 pair terminating in 3.5mm stereo > PR-2
Set on mic in for the 5V PIP
Looking through many pages of posts I see Gutbucket with 4060/4061 has been setting gain at 0, however others (possibly with other mics, I don't remember) note +15 or even +18. I'm finding with my home computer speakers (granted, much quieter than PA at show) I need at least +18 gain. I know Gut is an expert on these matters so I'm a little confused as to how one could get reasonable levels at 0 gain. Am I missing something?
I like the triple-press locking mechanism. I don't like the unlinked R/L gains... strange.
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
I'm not sure how to phrase this without sounding stupid, but if I set the gain to zero, then I don't see any levels on the screen for right and left channels i.e. they must be below -40. It seems I'm missing something big here.
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I have only used the recorder once and kept the gain at zero. After processing the gain in Adobe Audition I noticed some noise.
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I have only used the recorder once and kept the gain at zero. After processing the gain in Adobe Audition I noticed some noise.
Were you able to see levels on the screen? What mics were you using?
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
Is this also true for 24bit or only for 32bfp?
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I have only used the recorder once and kept the gain at zero. After processing the gain in Adobe Audition I noticed some noise.
Were you able to see levels on the screen? What mics were you using?
Didn’t check the levels in the screen and used DPA 4061 mics with the 5 volts PIP
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Setting it to mic in automatically provides the 5V PIP, correct? I didn't see any separate way to set it.
If you set the gain at 0, and no levels are seen while recording a rock show, then there's no way to visually confirm you're recording anything, and there would be a TON of noise in post.
I have only used the recorder once and kept the gain at zero. After processing the gain in Adobe Audition I noticed some noise.
Were you able to see levels on the screen? What mics were you using?
Didn’t check the levels in the screen and used DPA 4061 mics with the 5 volts PIP
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Manual says: Mic Plug-In Power 5V / Line In Switchable
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
Is this also true for 24bit or only for 32bfp?
There is no analog gain at either bit depth.
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
But if I run gain at 0 and get an extremely low volume recording I need to apply digital gain in post anyway...?
Please, someone explain it to me like I'm 5:
I'm plugging my DPA 4061s terminating in 3.5mm straight into "in" jack on PR-2. Input = mic (not line). Auto Gain off. Gain L=0 / R=0. Fresh batteries (tried with both Deity-supplied lithiums and regular alkalines). Turned up my speakers as loud as they go and hit record holding the mics close to them. While recording I see levels on PR-2 screen no higher than -40. Resulting file is very low volume which needs to be cranked up in post to be usable. Can someone point out what I'm doing incorrectly? Or is my unit defective and not supplying the 5V plug in power?
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There’s no difference between normalizing in post or running louder on the recorder, except that you can’t go over if you do it in post. If you need louder going in, you’d want to use a preamp.
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The only reason I got the PR-2 was to eliminate the need for a battery box or preamp. (I was the one who started the thread "CA-UGLY bit the dust, need another preamp solution for DPA 4061s" over in the battery boxes, preamps, etc. subforum.)
There’s no difference between normalizing in post or running louder on the recorder, except that you can’t go over if you do it in post. If you need louder going in, you’d want to use a preamp.
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Maybe get to a louder spot for your recording?
You absolutely can turn the gain up on the pr2, it’s just not going to lower the noise floor vs doing it in post.
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Or try more sensitive mics...
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
Is this also true for 24bit or only for 32bfp?
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The above chart seems to confuse bit depth and sampling frequency characteristics.
Bit depth dictates dynamic range.
Sampling frequency dictates resolution.
Perhaps I am taking this out of context? Can you provide link to your chart, please?
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Update:
Tested above referenced configuration from front row of medium-loud rock show. No gain setting issues with much higher SPL than at home. I ended up setting the PR-2 gain to +3 and was happy to see levels bouncing around.
What threw me off before:
Previously for these mics (DPA 4061) I always used a 9V battery box whereas PR-2 supplies only 5V plugin power. It didn't occur to me there must be a linear relationship between voltage and response. So the response was markedly different from what I've seen before testing other setups at home (speakers, claps, etc).
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The above chart seems to confuse bit depth and sampling frequency characteristics.
Bit depth dictates dynamic range.
Sampling frequency dictates resolution.
Perhaps I am taking this out of context? Can you provide link to your chart, please?
The charts show the difference between linear samples and floating point. With floating point, very low volume samples can be stored much more accurately than with linear samples.
I don’t know where these charts came from, but here you can find a nice explanation of 32bfp vs linear:
https://www.sounddevices.com/32-bit-float-files-explained/ (https://www.sounddevices.com/32-bit-float-files-explained/)
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The above chart seems to confuse bit depth and sampling frequency characteristics.
Bit depth dictates dynamic range.
Sampling frequency dictates resolution.
Perhaps I am taking this out of context? Can you provide link to your chart, please?
The charts show the difference between linear samples and floating point. With floating point, very low volume samples can be stored much more accurately than with linear samples.
I don’t know where these charts came from, but here you can find a nice explanation of 32bfp vs linear:
https://www.sounddevices.com/32-bit-float-files-explained/ (https://www.sounddevices.com/32-bit-float-files-explained/)
I am well versed in the differences between 24bit and 32bit fp and how to use. My question is in the claim that 32bfp provides "higher resolution". It does not. The advantage is in dynamic range only.
It would be useful to see if I am missing some context by not being able to see the full explanation within that snip.
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^ I think you are correct. Nyquist-Shannon and whatnot...
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The above chart seems to confuse bit depth and sampling frequency characteristics.
Bit depth dictates dynamic range.
Sampling frequency dictates resolution.
Perhaps I am taking this out of context? Can you provide link to your chart, please?
The charts show the difference between linear samples and floating point. With floating point, very low volume samples can be stored much more accurately than with linear samples.
I don’t know where these charts came from, but here you can find a nice explanation of 32bfp vs linear:
https://www.sounddevices.com/32-bit-float-files-explained/ (https://www.sounddevices.com/32-bit-float-files-explained/)
. My question is in the claim that 32bfp provides "higher resolution". It does not. The advantage is in dynamic range only.
I tend to disagree… Consider very very quiet parts in your recording. When using 16 or 24 bit linear storage, you’d be using just a handful of these 16 or 24 bits for these samples, so you’ll have relatively big quantization error compared to loud samples. Then, if you’d amplify this quiet audio, you will amplify the quantization error.
In case of 32bfp, even very quiet samples will use plenty of the available bits, so much less quantization errors compared to linear storage. So quiet samples can be amplified without amplifying the quantization error! That’s what’s being shown in the graphs…
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I tend to disagree… Consider very very quiet parts in your recording. When using 16 or 24 bit linear storage, you’d be using just a handful of these 16 or 24 bits, so you’ll have relatively big quantization error compared to loud samples. Then, if you’d amplify this quiet audio, you will amplify the quantization error.
In case of 32bfp, even very quiet samples will use plenty of the available bits, so much less quantization errors compared to linear storage. So quiet samples can be amplified without amplifying the quantization error! That’s what’s being shown in the graphs…
Do you have any examples of this happening at live music volume levels?
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I tend to disagree… Consider very very quiet parts in your recording. When using 16 or 24 bit linear storage, you’d be using just a handful of these 16 or 24 bits, so you’ll have relatively big quantization error compared to loud samples. Then, if you’d amplify this quiet audio, you will amplify the quantization error.
In case of 32bfp, even very quiet samples will use plenty of the available bits, so much less quantization errors compared to linear storage. So quiet samples can be amplified without amplifying the quantization error! That’s what’s being shown in the graphs…
Do you have any examples of this happening at live music volume levels?
Unfortunately I’ve had to deal with recordings with both very loud and very quiet songs, and on top of that a very conservative record level because of the unpredictable nature of the performance. So yes, a lot of dynamic processing was needed to make it a listenable recording. Likely the quantization errors are negligible compared to the noise in the quiet parts, but that’s not the point here. I was just trying to explain what was shown in the graph. I think it’s nice not having to worry about amplifying quantization errors during post processing, and I do regard that as an additional benefit of floating point storage, besides the larger dynamic range.
And even loud recordings will have sampled close to 0. It would be nice if these were stored with the same accuracy as high value samples.
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I’d just like to see a sample that demonstrates what you’re describing. I can see how trying to remove the dynamics from a recording can result in errors, however.
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The dynamic range of most shows probably doesn't exceed 60 dB. Maybe 80 dB if you're lucky. Even with a conservative level setting, 24-bit (or 16-bit for that matter) should cover it...
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Some amazing sounding recordings were made in the 80s using 14 bit converters and Betamax video recorders.
I still think this argument isn’t really relevant for the pr2 because the limitation is the analog stage not the digital stage. When it’s the right recorder and microphone combination it works and sounds great. If you overload the input or have a really super quiet source it might not be the right device, but it’s not because of 24 or 32 bits.
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Some amazing sounding recordings were made in the 80s using 14 bit converters and Betamax video recorders.
I still think this argument isn’t really relevant for the pr2 because the limitation is the analog stage not the digital stage. When it’s the right recorder and microphone combination it works and sounds great. If you overload the input or have a really super quiet source it might not be the right device, but it’s not because of 24 or 32 bits.
I don’t think there is even an argument :D. Somebody posted a graph trying to explain something, somebody else misunderstood the graph, I tried to explain what the graph is trying to show.
No need to do the 24bit vs 32bfp discussion here. Plenty of other threads for that!
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The dynamic range of most shows probably doesn't exceed 60 dB. Maybe 80 dB if you're lucky. Even with a conservative level setting, 24-bit (or 16-bit for that matter) should cover it...
Yeah, you can run a experiment like this:
- Open Audacity and record 10 seconds of audio with your microphone.
- Select all of it and click Edit > Remove Special > Silence Audio
- Export 3 files: 16-bit, 24-bit and 32-bit float.
- Open each file and start adding gain.
This is what I found:
16-bit
0 dB: silence
+10 dB: silence
+15 dB: barely perceptible noise
+20 dB: perceptible noise
+71.224 dB: quantization errors normalised to 0 dBFS
24-bit
0 dB: silence
+50 dB: silence
+60 dB: noise on the edge of perception
+65dB: barely perceptible noise
+70 dB: perceptible noise
+118.474 dB: quantization errors normalised to 0 dBFS
32-bit float
You can amplify it by any number and there will be no quantization errors
Now, if I take one of the quietest passages of music I've ever recorded (a very quiet piano intro), it was around 42 dB quieter than the loudest peaks in the recording (applause at -3 dBFS). If I isolate that part and normalise it to 0 dB, the 24-bit quantization errors will be at least 15 dB quieter than my ears can perceive at full volume with my specific combo of soundcard + headphones - and that's MUCH quieter than the recorder and microphone self-noise, which can be heard with much less digital amplification.
So I agree that quantization errors, if you're using 24-bit files, are negligible. You really need to fuck up your gain staging terribly for it to be relevant. 32-bit float devices are good at reducing noise in practice because they are optimised against their preamp/ADC self-noise. You can much more easily fuck up a recording by bumping against the recorder's analogue self-noise (by feeding too quiet a signal to a preamp/ADC combo that isn't so great) than against the file format's inherent noise (if using 24-bit - it's quite easy to do that with 16-bit).
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Just got back to town and following up. If I'm handing my DPA 4061>PR-2 rig off to someone else to run I just leave gain set to 0dB, and set the PR-2 to automatically start recording and then lock itself upon power up. Those situations are ones in which I know the noise floor of the venue will exceed that of the recording chain regardless of how gain is set. Just add gain later as needed later. No worries that way.
In my initial uses, the hot output of DPA Core 4060 was overloading the input stage of the PR-2 even when was set to 0dB gain. 4061 is significantly less sensitive and can accommodate the use of some input gain on the PR-2 without problems in all conditions I've run it so far, but that's not included anything ear-bleedingly loud.
When running the same rig myself I'll add some gain just to get the levels up to something reasonable depending on how loud the material is.. while still leaving plenty of headroom because the meters are VU (averaged) instead of PPM (peak). Makes the resulting recording more usable without processing, especially if playing back directly from the recorder, which is awkward with the PR-2 but I still do it. Using the same rig to record some acoustic guitar on the back porch a while back I bumped up the gain up significantly. I wish Deity would change the metering from VU (average) to PPM (peak) with a firmware update, or to combo meters, or make them menu switchable.
That's my practical take. I've not analyzed noise floor or anything yet, but haven't felt the need to as I've only used this rig for shows with high ambient noise floors and have not yet used it for classical recording in a very quiet hall
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Beyond the practical take.. (disclaimer, I'm not a digital audio EE!)
I think those staircase waveform images are one of the biggest deciets in digital audio. First off, a lot of folks, although not so much TS members, seem to believe that staircase represents the shape of the audio output. But as most here know, the output is always smoothed to a curve. The question is how accurately the shape of the smoothed output curve reflects the input. Sampling Theorem tells us that within a given bandwidth, the number of samples needed to do that needs only to exceed that bandwidth by two times in order to be able to retrieve the complete waveform, including all values between those sample times. As long as there are two or more sample points per cycle a steady state sine wave can be fully reconstructed. Extending that, a sinc function of overlapping sine waves is further capable of recreating a complex waveform shape that varies with time.
The devil is in the details of how the waveform is decimated and recreated, the filtering to limit the signal to within the usable bandwith, not in the bit depth and sample rate itself.
It would be alot less misleading if those images showed just the individual sample points at each "step corner" and not the lines connecting them that visually form the stair step representation. The actual sample points do not get connected by straight lines, they determine a series of overlapping curves that integrate to form the curved output waveform. The overlapping curves and averaging fit output waveform to the sample points. There are no stair steps.
It would also be less misleading if such images varied the spacing between sample points along the vertical axis with change of bit-depth, and the spacing between points along the horizontal axis with change of sample-rate. But such images almost never show that. Instead they are almost always drawn showing equal spacing along both axes.. as perfectly square stair-shaped steps.
The actual bandwidth limit of a recording is most likely to be solely determined by the dynamic range of the acoustic situation, beyond that by the dynamic range capability of the microphones, and beyond that by the preamp stage or ADC. A more complicated ADC arrangement designed to switch gracefully between multiple ADCs can extend the ADC range constraint, yet the range of a single 24-bit ADC most likely already exceeds that of the acoustic environment and most microphones. Within the bandwidth limits determined by the recording chain, the information represented in a 24-bit representation is going to be the same as in a 32-bit float representation. The 32-bit float storage representation just allows the 24-bit chunk of meaningful data to be shifted up and down as needed in the digital realm, it doesn't provide greater resolution within that meaningful range.
Remember, multiple switching ADCs and 32-bit float representation of the data are two different things. We can have one without the other, which manufacturers tend to gloss that over. A multiple switching ADC scheme extends the dynamic range envelope allowing for more lax real-world input gain setting by the user.. or no setting at all. But in simple terms, the "resolution" within that envelope is defined by the sample rate and determines the high frequency limit of the system, not the reproduction accuracy within the limits of the frequency range. Higher resolution extends the frequency range, but doesn't increase accuracy within the range.
a 32-bit floating point representation of the output from a non-dithered single 16-bit ADC contains the same information as a 16-bit fixed point representation of it and vice-versa. The 32-bit floating point digital container itself is vastly larger than the 16-bit digital container, making it capable of representing a far wider dynamic range, but in this case the useful data being output from the single 16-bit ADC and stored inside it remains the same.
Likewise, the electrical noise floor of a 32-bit float recorder is going to be determined by the analog input stage and ADC of the recorder, not by the data storage format.
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Honestly I am too impatient to read everything from the last few days, but for what it is worth, for myself, I run the deck only at 24 bit, I have it set for 5v pip, and I have the gain at +15 or +18 depending on the source which is usually rock, jam, reggae, solo, whatever. At 24 bit the gain applied is very important. If I went at 0db gain, I'd have lots of noise. It is no different than any 24 bit recorder in terms of setting gain appropriate to the source, except a preamp is not needed. The recordings at +15 or +18 sound great! I use a Church CA11 mostly and sometimes an AT 853.
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Honestly I am too impatient to read everything from the last few days, but for what it is worth, for myself, I run the deck only at 24 bit, I have it set for 5v pip, and I have the gain at +15 or +18 depending on the source which is usually rock, jam, reggae, solo, whatever. At 24 bit the gain applied is very important. If I went at 0db gain, I'd have lots of noise. It is no different than any 24 bit recorder in terms of setting gain appropriate to the source, except a preamp is not needed. The recordings at +15 or +18 sound great! I use a Church CA11 mostly and sometimes an AT 853.
The gain applied is very important if it is analogue gain. I think it's been said here the PR-2 has no analogue gain, but that sounds weird to me - it'd limit the use of the recorder too much. I'm not saying this is not true, just imho a weird decision.
To use a recorder I'm familiar with as an example, the Roland R-05's converter stage noise is -98 dBFS(A) (https://www.b4net.dk/posts/roland-r-05-technical-review/). If it didn't have an internal preamp at all, you'd depend exclusively on the microphone's sensitivity and the loudness of the source to produce a signal as far away from the ADC's self-noise as possible, and the recorder's EIN would be a fixed -98 dB. From experience, using microphones with fairly low sensitivity like the CA-11s, you'd struggle to reach -15 dBFS with zero gain even at the loudest shows (in compliance with UK law). A quiet solo singer-songwriter at a concert hall would very easily be recorded at -50 dBFS with no gain, which when normalised - either in post or by adding digital gain on the recorder - would bring that converter stage noise up to -48 dBFS! Which you can most definitely hear.
The way these recorders attain better EIN is by having analogue gain that increases the signal at a much higher rate than it adds noise (i.e. it might make the signal 10 dB louder while only adding 1 dB of noise). So instead of thinking of the Roland R-05 as a recorder that can reach an EIN of -121.8 dBu(A), I like to think of it as a recorder that gives you clear enough gain to increase the level of the microphone signal so that it can stay as far away from that -98 dBFS(A) noise floor as possible. Or you can use an external preamp for that. Either way, the ADC's noise floor is absolute, you cannot escape it.
Well, you can escape it, but only by having more than one of them calibrated differently, which the PR-2 doesn't have for stereo sources.
My conclusion here - IF it is true that the PR-2 provides no analogue gain at all - is that it is not fit for use with low sensitivity microphones for any show that isn't at least a moderately loud rock concert. Having to rely on the sensitivity of the mics alone doesn't give you a lot of flexibility unless you add a preamp to the chain, which then cancels out the advantage of it providing 5V PiP.
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So other recorders we're more familiar with, such as Sony M10, A10 etc, apply analog gain whereas the PR-2 only applies digital gain? What would be the rationale for doing so?
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The reason to run gain 0 is the gain is all digital so there’s no benefit to running higher than that.
Not that I wasn't to crap all over this, but is there anything documented somewhere showing this or is this based strictly on experience? Just picked a couple of them up not long ago but haven't opened them up yet to try myself.
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I thought I’ve read it but I’m not sure where. it’s what my testing suggests. There’s probably an analog gain stage but the slider at least appears to only change the digital.
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Beyond the practical take.. (disclaimer, I'm not a digital audio EE!)
I think those staircase waveform images are one of the biggest deciets in digital audio. First off, a lot of folks, although not so much TS members, seem to believe that staircase represents the shape of the audio output. But as most here know, the output is always smoothed to a curve. The question is how accurately the shape of the smoothed output curve reflects the input. Sampling Theorem tells us that within a given bandwidth, the number of samples needed to do that needs only to exceed that bandwidth by two times in order to be able to retrieve the complete waveform, including all values between those sample times. As long as there are two or more sample points per cycle a steady state sine wave can be fully reconstructed. Extending that, a sinc function of overlapping sine waves is further capable of recreating a complex waveform shape that varies with time.
The devil is in the details of how the waveform is decimated and recreated, the filtering to limit the signal to within the usable bandwith, not in the bit depth and sample rate itself.
It would be alot less misleading if those images showed just the individual sample points at each "step corner" and not the lines connecting them that visually form the stair step representation. The actual sample points do not get connected by straight lines, they determine a series of overlapping curves that integrate to form the curved output waveform. The overlapping curves and averaging fit output waveform to the sample points. There are no stair steps.
It would also be less misleading if such images varied the spacing between sample points along the vertical axis with change of bit-depth, and the spacing between points along the horizontal axis with change of sample-rate. But such images almost never show that. Instead they are almost always drawn showing equal spacing along both axes.. as perfectly square stair-shaped steps.
The actual bandwidth limit of a recording is most likely to be solely determined by the dynamic range of the acoustic situation, beyond that by the dynamic range capability of the microphones, and beyond that by the preamp stage or ADC. A more complicated ADC arrangement designed to switch gracefully between multiple ADCs can extend the ADC range constraint, yet the range of a single 24-bit ADC most likely already exceeds that of the acoustic environment and most microphones. Within the bandwidth limits determined by the recording chain, the information represented in a 24-bit representation is going to be the same as in a 32-bit float representation. The 32-bit float storage representation just allows the 24-bit chunk of meaningful data to be shifted up and down as needed in the digital realm, it doesn't provide greater resolution within that meaningful range.
Remember, multiple switching ADCs and 32-bit float representation of the data are two different things. We can have one without the other, which manufacturers tend to gloss that over. A multiple switching ADC scheme extends the dynamic range envelope allowing for more lax real-world input gain setting by the user.. or no setting at all. But in simple terms, the "resolution" within that envelope is defined by the sample rate and determines the high frequency limit of the system, not the reproduction accuracy within the limits of the frequency range. Higher resolution extends the frequency range, but doesn't increase accuracy within the range.
a 32-bit floating point representation of the output from a non-dithered single 16-bit ADC contains the same information as a 16-bit fixed point representation of it and vice-versa. The 32-bit floating point digital container itself is vastly larger than the 16-bit digital container, making it capable of representing a far wider dynamic range, but in this case the useful data being output from the single 16-bit ADC and stored inside it remains the same.
Likewise, the electrical noise floor of a 32-bit float recorder is going to be determined by the analog input stage and ADC of the recorder, not by the data storage format.
Thanks Gutbucket for your elaborated take on this. Things like resolution and accuracy can be confusing in the (digital) audio world.
IMHO, the practical benefit of the 32bfp format may be limited, except when 'whatever signal processing' is being done which could make the samples go over 0dB, in which case it's essential!
However, again IMHO, I think that if any digital signal processing is done on the output of one or more ADC's in a recorder (either amplification, filtering, dynamics processing, limiting, combining multiple ADC outputs, whatever), information/accuracy will be lost when storing in 24bits linear. This is particularly true for samples close to 0, where just the lowest couple of bits of the sample contain actual information if 16/24 linear storage would be used. Whether this is pure theoretical or practical: For my peace of mind, and if my recorder supports it, I would use 32bfp storage. (Unless that would turn the device into a mono recorder, as is unfortunately the case for the Deity PR-2 :( ) Personally I don't mind the 33% extra file size.
Apart from all this, I do see mixed reports here about the gain behavior of the Deity... Some say they get noisy recordings when gain set to 0, others say the gain is fully digital, so there is no need to set the gain to anything other than 0... I guess it would be good to get that sorted out...
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The way these recorders attain better EIN is by having analogue gain that increases the signal at a much higher rate than it adds noise (i.e. it might make the signal 10 dB louder while only adding 1 dB of noise). So instead of thinking of the Roland R-05 as a recorder that can reach an EIN of -121.8 dBu(A), I like to think of it as a recorder that gives you clear enough gain to increase the level of the microphone signal so that it can stay as far away from that -98 dBFS(A) noise floor as possible. Or you can use an external preamp for that. Either way, the ADC's noise floor is absolute, you cannot escape it.
EIN is pretty much a marketing tool. It is usually measured at really high gain settings (sometimes in the 50's to 60's dB - that's amplification by a factor of 1,000). The performance of an amplifier at these gain levels does not say anything about its performance at considerably lower gain levels that you would use when recording live music. On top of that, those settings are usually "A-weighted". This means they are basically applying an aggressive EQ curve that further reduces a lot of the audible frequency spectrum before measurement - including noise.
Also, where does the rumor about the PR-2's gain stage being all digital come from? This sounds really odd and I really can't believe it.
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My speculation as I've not tested anything, but there is at least an analog input stage that provides some sort of gain matching to buffer and provide the appropriate level to the ADC. Not sure if the "input gain" adjustment effects that or if its fixed and the gain adjustment is done digitally, but I imagine that at the very least the switching between mic-in and line-in sensitivity is managed in that analog gain stage. Would be nice if PIP could be switched on when in the less sensitive line-in mode to accommodate hotter output mics like the DPA 4060s I initially tried to use into it.
Firmware update wishlist:
>Change metering to PPM or combo VU/PPM (or make menu switchable).
>Option to link both channels of input gain adjustment when in 2-channel mode.
>Switchable PIP on line-in.
>Option to slave mulitple PR-2s together with clock and transport sync via a simple wired USB connection (rather than via separate wirelessly timecode gen box).
Haven't tried the last one, may be possible already.
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Apart from all this, I do see mixed reports here about the gain behavior of the Deity... Some say they get noisy recordings when gain set to 0, others say the gain is fully digital, so there is no need to set the gain to anything other than 0... I guess it would be good to get that sorted out...
I can tell you that it is sorted out. If at 24bit (with a live recording) you run gain at 0db you will get a crappy recording. If you set the gain the same as you would running 24 bit on any other machine , you will get as good a recording as you would get running 24 bit anywhere else.
32 bit is a different story and a mono recording with this deck. Many of the above comments do not specify what bit rate is being used. I am specifying and my data is based on my own results, so feel free to disagree, but if you have no personal experience I think commenting might just add confusion. If you choose to run 24 bit at 0db good luck, I hope it is a really really loud show. If someone is of the belief that there is no difference in 0db and +18db because it is digital, my advice is then run at +18. That way you can keep your belief and not get a crappy recording. ::) ;D
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
It does sound good. I'll still continue to record my way, but this is a nice recording!
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
It does sound good. I'll still continue to record my way, but this is a nice recording!
Thanks. The PR2 and KK14s make a REALLY easy rig to get in the door and pull the heat.
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I hate to stir this pot after it seems settled, but I just did a short experiment:
Church CA-11 microphones into the PR-2 (5V PIP), placed in front of a speaker. I played the intro to the song Aqualung by Jethro Tull at 95 dB (C weighted on REW with a calibrated mic)
So - it's loud, but not really quite up to concert level, which I've read is about 100 dB or so.
Recorded on the Deity in stereo, 24 bit, with the recording levels set at 0 dB, 15 dB, 24 dB, and 36 dB.
Sucked them into Audacity 3.7.1 and used Effect > Amplify to normalize each file.
Audacity added 34 dB to the 0 dB recording, 19.7 dB to the 15 dB recording, 10.6 dB to the 24 dB recording, and 0 dB to the 36 dB recording (this last one had clipped in two spots)
I then aligned the tracks and looped a "silent" section of the music near the start of the song. (the Aqualung song (from the "Special Edition" album) was played from Spotify on my phone (Galaxy S10e) and cast to a Chromecast Audio attached to my receiver)
Are you still with me?
The silent section of the track recorded at 0 dB sounded like shit. Like a bad cassette recording.
The track recorded at 15 dB was also noisy, but much quieter.
The tracks recorded at 24 dB and 36 dB were quieter still and sounded identical to each other with only a very slight amount of hiss. I think this hiss represents the sum of all the residual noise in the entire signal chain.
It is not clear to me where the noise in the 0 dB and 15 dB recordings is coming from. It's either inherent the Deity file, or Audacity adds it when boosting the volume.
TL:DR - For best results, please adjust the recording levels on the Deity PR-2 like you would on any other recorder.
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
I don't want to be unkind here because I think it sounds really good otherwise, but if I had made this recording, I'd be really bummed out that such a good capture had so much hiss. When I tape really quiet shows and then try to apply some sort of compression, I can often hear the hiss surfacing, disappearing and resurfacing. I try to go easy on the compression in that case. But I listened to "Ain't Got a Place" from your recording, and not only can I hear quite a bit of hiss throughout the entire file, but I can also tell the exact points at which the compressor was working due to the hiss getting louder or quieter (but never transparent).
I'm running the risk of sounding like a fool here if you go and say you didn't actually use a compressor and that I'm hearing things. But I don't think I'm wrong, it just sounds that clear to me.
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Dig McMurtry, will listen later..
To state the obvious, optimal gain required will depend not only SPL but also the sensitivity of the mics being used. Here is the sensitivity of some of the mics we've been discussing, arranged from low to high-
>Church Audio CA-11 (not sure)
>Neumann specs KK14 cardioid @ 3.6 mV/Pa ≅ -49 dBV (1 kHz into 1 kohm)
>DPA specs 4061 omni @ 6 mV/Pa; -44 dB re. 1 V/Pa (±3 dB at 1 kHz)
>AT specs 853 cardioid @ 11.2 mV/Pa; -39 dB re 1V at 1 Pa (standard 3-wire config, but I think lower that than with the 2-wire 4k mod typical around TS)
>DPA specs 4060 omni @ 20 mV/Pa; -34 dB re. 1 V/Pa (±3 dB at 1 kHz) TOO HOT!
As mentioned, in my use DPA 4060 @ 20mV/Pa proved too hot for concert recording into PR-2, while 4061 @ 6 mV/Pa seems about right with the potential to accommodate high SPL content.
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It is not clear to me where the noise in the 0 dB and 15 dB recordings is coming from. It's either inherent the Deity file, or Audacity adds it when boosting the volume.
It's coming from the amplifier. And, like every other analog amplifier, its equivalent input noise is better with higher gain settings and worse with lower gain settings. Also, your Church mics might have contributed some amount of noise. It's usually best to conduct noise tests without real microphones. Instead, you can connect a dummy load resistor to the recorder's input.
I did my own measurements (https://taperssection.com/index.php?topic=202228.msg2419577#msg2419577) some time ago, although I must admit I only tested it at max. gain to verify Deity's too-good-to-be-true EIN claims. I should probably repeat these tests with zero gain sometime...
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Dig McMurtry, will listen later..
To state the obvious, optimal gain required will depend not only SPL but also the sensitivity of the mics being used. Here is the sensitivity of some of the mics we've been discussing, arranged from low to high-
>Church Audio CA-11 (not sure)
My CA-11s are 19 dB less sensitive than my Clippys with Primo EM272M capsules. The official sensitivity for these capsules is -28 dB (±3dB at 1kHz, 0dB=1V/Pa), but according to MicBooster their improved RF protection makes them 6dB less sensitive than the EM272Z1 capsules, which are also rated at -28 dB. So -34 dB? Which would place the CA-11s at -53 dB (or -47 dB, if we consider the official EM272M specs).
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
I don't want to be unkind here because I think it sounds really good otherwise, but if I had made this recording, I'd be really bummed out that such a good capture had so much hiss. When I tape really quiet shows and then try to apply some sort of compression, I can often hear the hiss surfacing, disappearing and resurfacing. I try to go easy on the compression in that case. But I listened to "Ain't Got a Place" from your recording, and not only can I hear quite a bit of hiss throughout the entire file, but I can also tell the exact points at which the compressor was working due to the hiss getting louder or quieter (but never transparent).
I'm running the risk of sounding like a fool here if you go and say you didn't actually use a compressor and that I'm hearing things. But I don't think I'm wrong, it just sounds that clear to me.
Don't worry, I'm not thin skinned. I put it up, I don't mind commentary. I did use some compression in raising the volume so that the applause and audience noise between songs were less intrusive. I do agree there's a bit of hiss, but to me, it doesn't distract from the music. I will run with more gain next time I run the PR2 to see if it makes a difference.
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
I don't want to be unkind here because I think it sounds really good otherwise, but if I had made this recording, I'd be really bummed out that such a good capture had so much hiss. When I tape really quiet shows and then try to apply some sort of compression, I can often hear the hiss surfacing, disappearing and resurfacing. I try to go easy on the compression in that case. But I listened to "Ain't Got a Place" from your recording, and not only can I hear quite a bit of hiss throughout the entire file, but I can also tell the exact points at which the compressor was working due to the hiss getting louder or quieter (but never transparent).
I'm running the risk of sounding like a fool here if you go and say you didn't actually use a compressor and that I'm hearing things. But I don't think I'm wrong, it just sounds that clear to me.
Don't worry, I'm not thin skinned. I put it up, I don't mind commentary. I did use some compression in raising the volume so that the applause and audience noise between songs were less intrusive. I do agree there's a bit of hiss, but to me, it doesn't distract from the music. I will run with more gain next time I run the PR2 to see if it makes a difference.
It will...
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I was a bit bored today and decided to take a look inside the PR-2, hoping to find out what kind of ADC this thing uses.
The PR-2 contains two PCBs that are stacked on top of each other. I was able to identify a few ICs, but a some of them had their top markings removed. What a dick move by Deity! What I found out was that the whole thing is run by an ARM Cortex M4 32-bit MCU as well as the fact that they are using pre-assembled modules for the wireless (Bluetooth) connectivity.
Also, there is a Renesas DA7218 (https://www.renesas.com/en/products/analog-products/audio-video/audio-codecs/da7218-ultra-low-power-stereo-codec-single-ended-headphone-driver) stereo codec/headphone driver IC! Section 9.8 of its datasheet (https://www.mouser.de/datasheet/2/698/REN_da7218_datasheet_3v6_DST_20240408-3076081.pdf) describes the digital audio format. Something that stood out to me was the following:
The internal serialized DAI [Digital Audio Interface] data is 24 bits wide. Serial data that is not 24 bits wide is either shortened or zero-filled at input to, or at output from, the DAI’s internal 24-bit data width. The serial data word length can be configured to be 16, 20, 24 or 32 bits wide using the dai_word_length register bits.
Let this sink in! The ADC does not support 32 bits of audio data natively. It can, however, output a 32 bit data stream. So are we getting a 32-bit signal at all when recording in mono?. I'm not sure why they went that route. There are 32 bit codecs on the market (https://www.mouser.de/c/semiconductors/interface-ics/interface-codecs/?resolution=32%20bit).
Now take a look at table 29 in section 9.2.2. The DA7218's ADC does not support a sample rate of 44.1 kHz at all. And I had already been wondering why it wasn't possible to choose 44.1 kHz.
Also, since there was some discussion: The DA7218 has analog amplifiers, but they can be digitally controlled.
Also, there is an as-of-yet unidentified IC chip on the right side of the same PCB. The markings are just barely visible. I *think* the first line says "OPA", so this could be an analog op-amp, but I'm not sure. It's too hard to see. The second line seems to end with "75" (again, I'm not sure).
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Thanks for the peek inside. How easily did it come apart? Did it just snap back together? If easy enough and unlikely to get damaged in a way that requires gluing back together I might upen mine up to see if I can better ID the unmarked chips for you. Any insight into the potential of somehow physically wiring two or three of these together to share transport control and clock?
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Thanks for the peek inside. How easily did it come apart? Did it just snap back together? If easy enough and unlikely to get damaged in a way that requires gluing back together I might upen mine up to see if I can better ID the unmarked chips for you. Any insight into the potential of somehow physically wiring two or three of these together to share transport control and clock?
It came apart easier than I thought. I found it best to start at the top (left and right of the input/output connectors). Also, there were 2-3 sport where they used a tiny bit of glue to hold it all together.
Theoretically, it should be possible to wire two units together. The audio codec IC has PINs for the word clock. However, since it is a BGA chip, its pins are on the bottom and not easily accessible. So you'd have to see where to get access to the clock signal elsewhere. and you'd need to cut the trace for that signal on your slave device to avoid mixing two separate clock signals. Personally I'd say this is not worth it. A DR-2d is roughly the size of two PR-2s and can already record two stereo tracks simultaneously.
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Thanks.
Yeah, in actuality the primary advantage of wiring two together for me would be eliminating the need to carry the external 4ch preamp required in my DR2d based rig, and beyond that the smaller plastic PR2 units are stealthier than DR2d which uses a metal housing and looks more like an audio recorder. Otherwise for recording 4 channels I remain content with the DR2d rig. That's all on the practical side of things.
To advance further on the technical side of things I'd need to wire three PR2s together to enable 5 or 6 channels, the use of which is a practically achievable simple extension of what I've been doing and has been a long time goal. 8 channels is the ultimate goal, but effective use of that would require tiny 5V PIP powered fig-8s mounted coincidently with the current mics to form 4 M/S pairs in place of the 4 mono channels.
/dreams
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Well, another question is whether or not the word clock signal can actually be driven over a cable (which is significantly longer and significantly more exposed to RFI than a PCB trace) without additional buffering.
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Wondered about that. Would probably keep the 2 or 3 units adjacent to each other or together in a single USB battery-like housing, although one application would have them spaced apart from each other, but not by more than about 10". This is all above my DIY electronics level of expertise and comfort zone though.
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Hey, sorry to pop in with a question that may have been covered earlier in the thread and I missed it when I scanned through the posts.
Can you plug a usb microphone into the usb-c port and record that signal instead?
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Now arrived! That's really small! :D
First real recording will be Pantera for me in february.
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^ Try it without the battery box on something loud. The built-in PIP is 5V. Not as much as 9V battery box but more than most any other PIP equipt recorder
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^ Try it without the battery box on something loud. The built-in PIP is 5V. Not as much as 9V battery box but more than most any other PIP equipt recorder
I don't have mod cable for my mics. Maybe later make one sometime, I just need to research.
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Gotcha. You'll be good to go as is.
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Did anyone manage to use is as USB audio interface? when I switched to that mode, the device manager in Win10 says unknown device. I can't find drivers for this.
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Looks like I have some catching up to do in this thread. I was popping in to mention that I tried a dual recording scenario last night. It my wives birthday so I wanted to travel light for recording. We went to Ryan Montbleau solo. I have two Deity PR-2's. I still haven't used them too much. Previously I wondered if we used two units with synced timecode, if it would be a viable four track recording method. Well, without much experimentation or testing, it appears no. Quick version is that I we sat in the balcony and I used AT853's to PR-2 (A) and had a sbd feed to PR-2 (B). I used the app to sync time code and let them roll for 3hrs. I wasn't really expecting this to "work as desired" in terms of having actual frame lock sync. It didn't. I aligned the recording at the top and there was definite drift by the end of the show. Compensated with time stretching pretty easily though.
Now, it's been a little while and I can't even recall what show, or event I did this for. But I did a successful version of this using the two PR-2's but one was running in TC mode with a TC cable feeding a Zoom F8. This configuration works.
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https://archive.org/details/james-mcmurtry-20240915/20240915+T01+-+Painting+by+Numbers.flac
acoustic show with a pa. Not a super loud show, and I recorded at 0 gain. kk14->pr2
I don't want to be unkind here because I think it sounds really good otherwise, but if I had made this recording, I'd be really bummed out that such a good capture had so much hiss. When I tape really quiet shows and then try to apply some sort of compression, I can often hear the hiss surfacing, disappearing and resurfacing. I try to go easy on the compression in that case. But I listened to "Ain't Got a Place" from your recording, and not only can I hear quite a bit of hiss throughout the entire file, but I can also tell the exact points at which the compressor was working due to the hiss getting louder or quieter (but never transparent).
I'm running the risk of sounding like a fool here if you go and say you didn't actually use a compressor and that I'm hearing things. But I don't think I'm wrong, it just sounds that clear to me.
Just chiming in to say I agree. It's a decent recording and enjoyable but it does sound like it was recorded on tape with this level of noise. The noise is fluctuating too. I'm guessing from compression or upward compression. I don't know how much the levels were adjusted but it obviously brought the noise floor up with it. In my little use with the recorder I am finding it can have a bit too much noise in 24bit mode and levels should be set appropriately to whatever degree possible.
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Looks like I have some catching up to do in this thread. I was popping in to mention that I tried a dual recording scenario last night. It my wives birthday so I wanted to travel light for recording. We went to Ryan Montbleau solo. I have two Deity PR-2's. I still haven't used them too much. Previously I wondered if we used two units with synced timecode, if it would be a viable four track recording method. Well, without much experimentation or testing, it appears no. Quick version is that I we sat in the balcony and I used AT853's to PR-2 (A) and had a sbd feed to PR-2 (B). I used the app to sync time code and let them roll for 3hrs. I wasn't really expecting this to "work as desired" in terms of having actual frame lock sync. It didn't. I aligned the recording at the top and there was definite drift by the end of the show. Compensated with time stretching pretty easily though.
Now, it's been a little while and I can't even recall what show, or event I did this for. But I did a successful version of this using the two PR-2's but one was running in TC mode with a TC cable feeding a Zoom F8. This configuration works.
Thanks for this, as I've been wondering but I still only have one PR-2.
Can you tell me more about connecting the PR-2 and F8 with a timecode cable? Direct connection? Which port? Didn't need to use a TC-1 box? One time jam sync at start or continuous?
Are you able to timecode two PR-2s together for continuous with a cable connection between them?
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Gotcha. You'll be good to go as is.
I have a pair of Panasonic WMA61A unmodded caps that I built many many years ago. They seems to work with the PIP, but gonne try them sometime with something loud, but they can't handle high SPL's, so I will try it something not really interesting.
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Can you tell me more about connecting the PR-2 and F8 with a timecode cable? Direct connection? Which port? Didn't need to use a TC-1 box? One time jam sync at start or continuous?
Are you able to timecode two PR-2s together for continuous with a cable connection between them?
I haven't even really scratched the surface and had only the one trial that I honestly can't even really remember. I used a 3.5mm TRS to BNC TC Cable. One Deity gets put into TC mode and in this mode acts like a TC-1 and cannot record. My Zoom F8 would have been set to accept external TC. I need to work with the workflow some more to be able to speak to it better. https://deitymic.com/blog/deity-pr-2-timecode-settings/
You know, it's possible I didn't "do it right" on Saturday. I should do some more testing before I can more confidently state it didn't work. I just fired up the devices and I see one was set to AUTO and one was set to ONCE. I didn't double check the units to try to ensure they were running in sync. In fact I used my phone to film the set, so the app was not accessible. I'm tinkering with it right now and the timecode seems to lock. That said, I didn't change anything and as I mentioned earlier, it was a 3hr recording and there appeared to be drift in post. I'll try to come up with a better test.
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Stereo monitoring and Bluetooth issue question
I'm new to the forum and I'm under big impression of users knowledge here.
I'm a field/nature recordist
I recently purchased PR2 (World version) and was amazed by it's size and weight.
I'm using Sennheiser HD25 and Beyer DT770 and Usi and EarSight Mics.
Questions:
- Is there a way to monitor stereo signal in stereo? I can't find a way to do that
- Do you have problems with Bluetooth pairing? I tried it with iphone 13 (latest ios), did BT reset, and I'm unable to connect. I even bought second one just to check if one I had was faulty. The same situation.
I would be grateful for your advice
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With the non-US version you should be able to switch on/off audio pass-through from the input jack to the headphone jack, and when switched 'on' it should pass signal to allow monitoring. Look for a little manual switch located between the input and output jacks.
Adjusting output level is a bit wonky as it can only be adjusted via the on-screen menu, in something like +/-3dB steps, and might not be accessible while recording. Don't think it is but I've not tried to do so with my US unit, however I can confirm that headphone monitoring level cannot be change during file playback. Must first stop playback, then navigate to the correct menu to change output level to the presumed correct setting (with no playback occuring), then restart playback fo the file again.
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Thank you Gutbucket, yes I can monitor with toggle to the left but only in mono.
Playback is in stereo though.
Did you have any problems with bt pairing?
With the non-US version you should be able to switch on/off audio pass-through from the input jack to the headphone jack, and when switched 'on' it should pass signal to allow monitoring. Look for a little manual switch located between the input and output jacks.
Adjusting output level is a bit wonky as it can only be adjusted via the on-screen menu, in something like +/-3dB steps, and might not be accessible while recording. Don't think it is but I've not tried to do so with my US unit, however I can confirm that headphone monitoring level cannot be change during file playback. Must first stop playback, then navigate to the correct menu to change output level to the presumed correct setting (with no playback occuring), then restart playback fo the file again.
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Haven't used bluetooth or the app with mine.
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I tried to encode the raw recordings to flac with TLH and it said the following error wit all files: file is truncated or otherwise corrupt.
The files play fine, and can be opened with Reaper without problems, no glitches.
Maybe something in their header that flac encoder does not likes?
Also tried to play with the includes foreign metadata option, but no luck.
I want to do this because I always compress my unedited files to flac to save space in the archives.
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Maybe try exporting the file from your DAW in .flac format? I use Audition and that is an option.
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So deity says there is analog gain with separate preamps for left and right channel. So I was wrong before
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Ran a test at warren haynes last night. Set 1 had one channel at 0 and 1 at +15db. I brought the level up in ozone.
https://archive.org/details/WHB20250207.kk14.pr2
Neumann KK14s -> PR2 balcony center
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For the BT pairing; when not in use I take the batteries out. When needed, I put the batteries in and pair it to my phone before I leave home and then switch off. When I get to the show and switch on and try pairing, it's fine. If I don't pair that first time at home, I have to go through all the BT setup at the show, which can be awkward. If that all makes sense.
- Do you have problems with Bluetooth pairing? I tried it with iphone 13 (latest ios), did BT reset, and I'm unable to connect. I even bought second one just to check if one I had was faulty. The same situation.
I would be grateful for your advice
[/quote]
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So deity says there is analog gain with separate preamps for left and right channel. So I was wrong before.
Makes sense. Haven't listened but does that equate to perceptible low level noise in the 0dB channel of the Warren Haynes recording?
The second part seems obvious since technically there must to be a separate preamp circuit for each channel. Presumably when operating in single channel 32-float mode Deity sets them them to different predetermined determined gains and switches between the two.
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Public alert warning!! concerning previously mentioned support gear-
I strongly recommend that anyone using the cheap Ebay 2x female Microdot > screw-locking stereo-Miniplug "Y" adapter for DPA mics which we discussed early in the thread, unscrew up the miniplug connector side and check the solder joints and ground/strain-relief crimping. The one I was using failed over the weekend - quite aggravating as I was really looking forward to recording several acts in quite oportune situations. Opened it up to find the ground/shield/strain-relief connection was crimped only and not soldered, and had worked loose allowing the fine conductors to pull and twist, breaking the solder-joints for the signal connections. I'd intended to open that up initially upon receipt to double check the solder-joint and crimping quality but was rushed at the time and never did it, and ended up suffering for it this past weekend after not that many uses.
The light blue wires of the Y adapter are also starting to delaminate a bit. The blue exterior jacket layer appears to be some sort of spiral plastic ribbon wrapped around a white internal jacket. I suspect its purpose is to make pulling cable easier in its intended application, and that it is not meant to be used as an external exposed jacket. Not a recording killer but unfortunate. I may just unwrap the blue layer and cut it away when I go in and repair the connections. Didn't notice that delamination earlier as I have both the Y adapter and mic wires sheathed in soft body-harness "quiet" techflex. I only noticed it when disassembling to diagnose the open circuit problem.
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I wanted to share my experience with the Deity PR-2. I used it to record a semi-acoustic gig last week, featuring both amplified and unplugged acoustic sections. Overall, the recording quality was impressive. The only noticeable drawback was the self-noise, which became apparent during the quieter parts. However, I believe the positives outweigh this issue.
The size of the PR-2 is perfect—it easily fits into the small coin pocket of my jeans, even with my AT-853 microphones plugged in (no screw locking needed). It remained securely attached throughout the recording, and I never had concerns about it coming loose. I connected the PR-2 to my iPhone 11, and the Bluetooth functionality worked flawlessly. The app consistently reconnected, even when I wasn't actively using it.
I set the gain to +21, and I saw DB peaks around 85 on my phone's app. With +21 gain, I had plenty of headroom in my DAW, ensuring I didn't have to worry about clipping. The PR-2 is ideal for those who prioritize portability and ease of use when bringing gear to a venue, as well as those who appreciate the Bluetooth feature. Also, the battery life is outstanding—lasting throughout the entire gig without any issues!
I love the PR-2 and look forward to using it for future recordings. A huge plus is that I won’t ever need a battery box since the 5V plug-in power (PIP) works perfectly!
You can listen to the recording here: https://e.pcloud.link/publink/show?code=kZ79WlZplmLSohoJHXrrqmxzWQ9rF8HrNok (https://e.pcloud.link/publink/show?code=kZ79WlZplmLSohoJHXrrqmxzWQ9rF8HrNok)
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Fun fact: The PR-2 fits perfectly in an empty playing card deck. Just some food for thought.
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The only noticeable drawback was the self-noise, which became apparent during the quieter parts. However, I believe the positives outweigh this issue.
Are you able to qualify this statement at all? Controlled and / or A/B testing? Point to specific examples of the self-noise being greater than the mics themselves / ambient room noise / etc.?
My understanding of specs / self-testing thus far has left me with the conclusion that unless the levels are set "erroneously" low, the self-noise of the mics / room a/c & heating / etc. has been the limiting factor (i.e. the self-noise of the recorder is eclipsed by other factors).
Any additional data / experience you can share?
Thanks in advance!
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The pr2 definitely has the highest self noise of any device I use. It’s plenty for most of the style recording we do, but ambient recordings etc will definitely show higher noise than is typical.
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Yes. Not a device I would use for field recordings, but still considering if I should get it for music.
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Ran into a situation last night where the PR-2 thought the microSD was full and stopped allowing me to record but the card was basically empty. There were no noticeable errors on the card.
Formatting the card fixed the problem though. Seemed to be an odd issue. Wonder if it was a one-off issue or a bug with the firmware.
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Hey all,
I wanted to pass along some details about a recently uploaded show from the Jon Anderson & The Band Geeks performance at the Rialto Theatre in Tucson (April 1, 2025). Although I’m not the one who recorded it, the uploader (Tapehead2) shared some important information regarding technical issues that affected his recording.
What Happened:
Tapehead2 recorded the performance with a Deity PR-2, but it turns out there were several digital “digiskips” throughout the show. It appears the problem stemmed from an incorrectly formatted memory card, resulting in multiple one-second skips across the recording.
You can check it out here: http://www.dimeadozen.org/torrents-details.php?id=791324 (http://www.dimeadozen.org/torrents-details.php?id=791324)
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Last night, I recorded eight hours of white noise using my AT853s with the PR-2, and I didn't notice any digital skips in the waveform. Everything worked great. I used a Sandisk Extreme MicroSDXC - 64GB Class 10.
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I'd love to challenge my PR2 to see whether I can get it to skip. I've done 4 shows without a problem so far (stereo, stock memory card, no bluetooth).
Is white noise appropriate as a test? If I cut 1 second out of that file and rejoined the pieces, could you find it?
I was thinking of recording for a couple of hours at a time (stopping before the auto split) with two independent recorders and comparing file lengths/times. They should match within milliseconds. If the PR2 is shorter by a second or two, that should indicate skipping. Would this be a good challenge?
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I’ve got to say, I’m really starting to feel confident using the PR-2—even for critical recordings. I taped an intimate in-store gig today by Dutch singer-songwriter Lucas Hamming, and the sound came out incredible! Have a listen: https://e.pcloud.link/publink/show?code=kZv8UdZR2Q4ezVuMTmVf7wpcAqOXJFqCgJy (https://e.pcloud.link/publink/show?code=kZv8UdZR2Q4ezVuMTmVf7wpcAqOXJFqCgJy)
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Impressive result! Very nice.
I barely tape acoustic shows but if I did I'd be taping in 32 bit float as well!
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Hi, I was the one referred to several posts ago re: skips in PR2 file. I'm not sure whether the issue stemmed from a bad format. In any case I recorded another show the other night with a new card, newly formatted, and did not experience any skips.
I wrote to Deity support and gave them samples of the skips. Waiting to hear back from them.
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I recorded a lot with it since I bought in january and haven't experienced any skips with that, I'm using with the included Kingston card.
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Any tips to how to use the included laverier mic properly to record speech? This one might be useful to me for shooting interviews with musicians.
I connected it, set input to mic, pushed up gain to +3dB and still faint sound got recorded while I clipped on my t-shirt and recorded some talking.
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The gain shouldn't matter in 32-bit float mode.
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New firmware out: https://deitymic.com/firmware/
Firmware V1.9 (LATEST)
– Fixes rare bugs
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I had written to Deity tech support about the skip issue; they advised me to update the firmware to V1.9. Not sure if they actually identified the bug and fixed it, or if this was simply a default response.... in any case, I've recorded one show since doing the update and no skips noted. Though, before the show that had skips in it I had recorded others without issues, so it'll be a while until I can safely say the problem is resolved.
New firmware out: https://deitymic.com/firmware/
Firmware V1.9 (LATEST)
– Fixes rare bugs
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Today, I taped an in-store gig using the Deity PR-2 and modded AT-mics, and it turned out pretty well.
👉 Listen to the recording: https://e.pcloud.link/publink/show?code=kZM12qZXhJXxsCVPluil1wmGFw3hfw5qjpk (https://e.pcloud.link/publink/show?code=kZM12qZXhJXxsCVPluil1wmGFw3hfw5qjpk)
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First of all, as this is my first post a "Hi" to all, and thanks for having me.
I try to make that as short as possible. As I switched away to a true RAW video camera app on my phone, this app cannot (like all other 3rd party apps) apply a negative gain, which will blow all my concert footage sound.
I am now looking to tape the sound for my footage in the best way I can afford right now.
Situation: Always front row, right at the barricade, in venues from 300 to approx. 5000 people, practically never outside, stealth is nearly as important as SQ in my case, sadly. That is why I came on the Deity through my research. It seems like the optimal device for my needs.
However, although I wish I could, two DPA 4060 (or the likes) are not achievable right now for me.
Given that I am always right in front of stage as said, do you think I can achieve proper results here if I just buy a second W.Lav Pro to record in stereo and make that my new to-go kit? I somehow saw on the official page that they only allow 110db SPL max (with a *), which might be too little in that case, somewhere else I read >130db which should be plenty.
Any experiences or advice for that specific situation would be highly appreciated.
Many thanks!
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^ How about the Countryman B3? It's about half of the price of the DPAs.
A 110 dBSPL max will be potentially problematic for loud rock shows, in my opinion. I don't think it goes above that much, but occasionally.
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^ How about the Countryman B3? It's about half of the price of the DPAs.
A 110 dBSPL max will be potentially problematic for loud rock shows, in my opinion. I don't think it goes above that much, but occasionally.
Absolutely, with 110db SPL max I think I would not buy them/trust them, especially so close up front, I am at a lot of metal/core concerts, amongst others, and this might be always right on the verge then... According to my dear ears some are over 110dB for sure, 130 though as said is another number, no worries there.
I might have to ask at Deity directly what that: *Some microdot adapters may allow for higher SPL amounts...means in reality
The countryman seem nice SPL-wise but they are also around 280,- EUR here a piece, which is still a lot right now, for me it would be preferable if I could stay somewhere under 200.- per mic. I know this might mean compromises but it is what it is.
Interesting for me as said would be also experiences with the combination, if there.
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^ How about the Countryman B3? It's about half of the price of the DPAs.
A 110 dBSPL max will be potentially problematic for loud rock shows, in my opinion. I don't think it goes above that much, but occasionally.
Absolutely, with 110db SPL max I think I would not buy them/trust them, especially so close up front, I am at a lot of metal/core concerts, amongst others, and this might be always right on the verge then... According to my dear ears some are over 110dB for sure, 130 though as said is another number, no worries there.
I might have to ask at Deity directly what that: *Some microdot adapters may allow for higher SPL amounts...means in reality
The countryman seem nice SPL-wise but they are also around 280,- EUR here a piece, which is still a lot right now, for me it would be preferable if I could stay somewhere under 200.- per mic. I know this might mean compromises but it is what it is.
Interesting for me as said would be also experiences with the combination, if there.
I'm selling a pair of Audio-Technica AT943 cards for €150! They work excellent with the PR-2. https://taperssection.com/index.php?topic=207252.0 (https://taperssection.com/index.php?topic=207252.0)
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[snip..]two DPA 4060 (or the likes) are not achievable right now for me.[snip]
4060 is not what you want anyway. It is capable of handling the sound pressure levels, but it's sensitivity is overly high for use into the PR2, resulting in distortion in concert recording situations. You'll want something more akin to 4061 instead. 4061 has a somewhat higher max SPL handling ability, but more importantly its sensitivity is low enough that it will not overload the PR2 as long as input gain is set correctly. The other mics mentioned are similar to 4061 in sensitivity.
Sensitivity of the miniature DPA omnis:
4060 = 20 mV/Pa; -44 dB re. 1 V/Pa
4061 = 6 mV/Pa; -44 dB re. 1 V/Pa
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[snip..]two DPA 4060 (or the likes) are not achievable right now for me.[snip]
4060 is not what you want anyway. It is capable of handling the sound pressure levels, but it's sensitivity is overly high for use into the PR2, resulting in distortion in concert recording situations. You'll want something more akin to 4061 instead. 4061 has a somewhat higher max SPL handling ability, but more importantly its sensitivity is low enough that it will not overload the PR2 as long as input gain is set correctly. The other mics mentioned are similar to 4061 in sensitivity.
Sensitivity of the miniature DPA omnis:
4060 = 20 mV/Pa; -44 dB re. 1 V/Pa
4061 = 6 mV/Pa; -44 dB re. 1 V/Pa
thx for the hint, but as I said 60 or 61 does not really matter much, as they are way out of price range for a pair, sadly even the Countryman B3 are out of question, I would need to get something where I can get away with lets say 200,- tops for a single mic, and that is already quite high, it's sadly otherwise not doable for me in the moment.
Honestly I thought that there has to be something for that money which is not a half-a..ed solution and suits my usual recording situation/placement.
I mean the W.Lav Pro seems to be a decent mic from all I read, esp. for the price, but I am in awe of the only 110dB SPL, and I think rightfully so in my case.
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I'm selling a pair of Audio-Technica AT943 cards for €150! They work excellent with the PR-2. https://taperssection.com/index.php?topic=207252.0 (https://taperssection.com/index.php?topic=207252.0)
Do you send from NL? Do they work under clothing? I know this is suboptimal per se, but I see no other way...For me this whole situation is totally awkward, because they are really strict here in the venues re recording equipment and as I am always front row this is will always be a high risk situation because it is not just about getting in.
I will also not being able to stand more or less still as
a. I am filming as said and
b. on some concerts there is really hell even in the front row, although it is normally easier than farther behind where the pit(s) start
It is so crazy that there is no proper way to apply a negative gain on phones. I mean there is, and the recordings are also pretty usable if properly post-processed, but then they record in 99% mono only...Sadly. All other solutions that "promise" gain reduction are just fraud because they just lower the gain post-input and therefore post-clip, so completely useless
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I'm selling a pair of Audio-Technica AT943 cards for €150! They work excellent with the PR-2. https://taperssection.com/index.php?topic=207252.0 (https://taperssection.com/index.php?topic=207252.0)
Do you send from NL? Do they work under clothing? I know this is suboptimal per se, but I see no other way...For me this whole situation is totally awkward, because they are really strict here in the venues re recording equipment and as I am always front row this is will always be a high risk situation because it is not just about getting in.
I will also not being able to stand more or less still as
a. I am filming as said and
b. on some concerts there is really hell even in the front row, although it is normally easier than farther behind where the pit(s) start
It is so crazy that there is no proper way to apply a negative gain on phones, I mean there is, and the recordings are also pretty good if properly post-processed, but then they record in 99% mono only...Sadly. All other solutions that "promise" gain reduction are just fraud because they just lower the gain post-input and therefore post-clip, so completely useless
Omnis are better when you're moving around a lot, since cardioid mics are directional—so I'm not sure if my mics will work well for your needs. But yes, I’m sending from the Netherlands. Here's a recording I made with the AT-943s from the balcony: https://e.pcloud.link/publink/show?code=kZ4aLdZRjesygCPgmScaki5Ssa2aFxF8iJV (https://e.pcloud.link/publink/show?code=kZ4aLdZRjesygCPgmScaki5Ssa2aFxF8iJV)
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Omnis are better when you're moving around a lot, since cardioid mics are directional—so I'm not sure if my mics will work well for your needs. But yes, I’m sending from the Netherlands. Here's a recording I made with the AT-943s from the balcony: https://e.pcloud.link/publink/show?code=kZ4aLdZRjesygCPgmScaki5Ssa2aFxF8iJV (https://e.pcloud.link/publink/show?code=kZ4aLdZRjesygCPgmScaki5Ssa2aFxF8iJV)
Oh yea I just saw that these are cardioids, I think this is not worth the hassle then. But thanks anyways.
I am constantly scratching my head on this whole situation, and I think even with more, or unlimited budget, I am in a pretty awful place. Best picture quality right now technically possible from a stealth perspective, but due to that literally f..ed with the sound, no matter how I put or twist it in my mind :shrug:
No one wants to see a concert vid with overly subpar sound, we have enough of that on YT and Insta already
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*purely hypothetical, I don't know anyone who ever ran a Multicam stealth video setup with stealth audio, because that would be bad*
Make friends, get the person recording audio into the sweet spot for audio, and get whatever cameras you have into the spots that make the most sense for them. Jam sync timecode if possible, but if not, you can sync in post.
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*purely hypothetical, I don't know anyone who ever ran a Multicam stealth video setup with stealth audio, because that would be bad*
Make friends, get the person recording audio into the sweet spot for audio, and get whatever cameras you have into the spots that make the most sense for them. Jam sync timecode if possible, but if not, you can sync in post.
Yea I think it will probably end up with a construct of the like, because honestly I played everything else through and through and through, and I was very considerate imo, but came to no solution that would be real practicable alone anymore. My problem is I am a bit of a chaser when I want to get something to work on my own. But one also has to be able to admit that not everything is doable alone, at least not if one expects a certain quality of the work he does himself, or delivers in the end.
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Posted this in the a10 thread….
I’m sure somewhere in here it talks about it but how is the a10 at not setting off metal detectors?
My m10 sets it off
Looking for something that will not, or at least not as much…
A10
Deity maybe
Looking to run 4.7 mod at853 > recorder for shows that I have no option in going back “to car”.
Ie a show coming up is 30 min train ride away
Tia
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Of all recorders currently available (other than using a phone to record along via a recording app), PR-2 is is least likely of any to set off a metal detector.
That said, metal detection will depend on how the sensitivity of the detector is set. I've seen some cranked up so high they detect as single small coin, even a small washer in a pocket. And others turned down so far they barely detect much of anything.
I just hold it above my head along with my phone and keys as they wand everything else down. No prob. If a walk-through you are paranoid about and don't trust, probably just put it under the phone, keys, wallet and any other stuff that goes in the basket. Its small and square and even with entry-level Jedi skills should be easy to pass off as a phone battery, pager, or remote of some kind if even noticed at all.
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..I've wondered, and asked here at TS, but have never definitively figured out if different AA battery types increase or decrease metal detectability when installed in the recorder. Regardless, if paranoid, might take the batteries out and carry them through separately.
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..I've wondered, and asked here at TS, but have never definitively figured out if different AA battery types increase or decrease metal detectability when installed in the recorder. Regardless, if paranoid, might take the batteries out and carry them through separately.
thanks
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Read through the entire thread, and while I was initially disappointed by the lack of 32 bit stereo, I’m going to give it a go with some at853s (4.7 mod) that I just picked up. Seems pretty straightforward…pip on, line in, set levels with the app…am I missing anything?
Had been recording with a mv88 > iPhone, but looking forward to a more proper rig.
Also, if anyone is looking to unload theirs…I’m interested.
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Of all recorders currently available (other than using a phone to record along via a recording app), PR-2 is is least likely of any to set off a metal detector.
That said, metal detection will depend on how the sensitivity of the detector is set. I've seen some cranked up so high they detect as single small coin, even a small washer in a pocket. And others turned down so far they barely detect much of anything.
I just hold it above my head along with my phone and keys as they wand everything else down. No prob. If a walk-through you are paranoid about and don't trust, probably just put it under the phone, keys, wallet and any other stuff that goes in the basket. Its small and square and even with entry-level Jedi skills should be easy to pass off as a phone battery, pager, or remote of some kind if even noticed at all.
I agree with you there. I've gone through many walk throughs with my Edirol-R9 and I did not set it off. But youre right, they all have different sensitivities. So yeah most times you can hold it in your hand now with all your other stuff and no one bothers to look. The mics are a bigger deal to me, that has set off detectors in the past so I use a belt clip mounted glasses case that has a hidden pocket that fits the wires, putting everything into the basket now is not a problem and with the PR-2, that can go into my new wallet. Battery box is now a car key FOB so I just walk through as they pass everything to me.
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I just picked up PR-2 and everything so far is working.....HOWEVER...I have 2 concerns.
I've updated it to the latest firmware but it will not connect to my PC via USB to transfer files...changed USB/System setting to Reader, tried different cables, computers, etc, but not even seen at all. Anyone seen this issue??
Also, Ive tested recording with my SP-CMC-8's, with and without the battery box and the levels seem pretty low. I have not actually tested in the wild but has anyone have very low levels on their mics where you have to turn the gain up real high ??
Thanks in advance.
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I just picked up PR-2 and everything so far is working.....HOWEVER...I have 2 concerns.
I've updated it to the latest firmware but it will not connect to my PC via USB to transfer files...changed USB/System setting to Reader, tried different cables, computers, etc, but not even seen at all. Anyone seen this issue??
Also, Ive tested recording with my SP-CMC-8's, with and without the battery box and the levels seem pretty low. I have not actually tested in the wild but has anyone have very low levels on their mics where you have to turn the gain up real high ??
Thanks in advance.
It’s pretty common with the SP-CMC-8s and similar AT mics with the Low Sensitivity Mod that you’ll need to boost the gain a bit higher than usual—around 18 to 24 dB is normal with the PR-2. This helps prevent clipping and keeps your recordings clean. The good news is you can always bring the levels up further in post-production if needed. Hope that helps!
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
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Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
For avoidance of doubt, can you explain/elaborate?
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Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
For avoidance of doubt, can you explain/elaborate?
Just ensure the peaks stay below -10dB, as they will effectively peak higher during playback or processing.
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It's a pseudo vu meter, instead of a peak meter. So if you're hitting -10db regularly, you're going to have spikes that will go much higher.
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Was typing while grawk answered..
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
For avoidance of doubt, can you explain/elaborate?
The metering is in VU (Voltage Units) and not PPM (Peak Program Meter). VU is averaged over a short-time window, emulating mechanical "needle" meters and human "as heard" levels, but its slower reaction time will not indicate the full extent of fast transient peaks. PPM indicates the actual peak values, and usually includes a slower peak hold indication in order better see them.
I really wish Diety would include an option to switch the metering to PPM, as avoiding overs is one of the most important goals in a recorder, and all other modern recorders that I know of use PPM (or include the option to choose).
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ah, thanks all for the explanation. That's pretty unintuitive, but not so bad as long as it's understood I guess.
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It’s pretty common with the SP-CMC-8s and similar AT mics with the Low Sensitivity Mod that you’ll need to boost the gain a bit higher than usual—around 18 to 24 dB is normal with the PR-2. This helps prevent clipping and keeps your recordings clean. The good news is you can always bring the levels up further in post-production if needed. Hope that helps!
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
Ahh right, that makes sense. Thanks for the reply. I've had these mics for nearly 10 years now and "I think" I had that mod done so that could be why. I guess my concern then is based on what people are saying that increasing the gain too much on the PR-2 is causing inherent hiss/noise to the recording??
I guess I will need to go to a concert this weekend and test all this out to see if that is an issue.
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the problem is the pr2 analog stage is somewhat noisy, so quieter sources will have more noise on a pr2 than other devices because the gain is happening digitally after the noise is in the signal path.
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the problem is the pr2 analog stage is somewhat noisy, so quieter sources will have more noise on a pr2 than other devices because the gain is happening digitally after the noise is in the signal path.
Hmmm, thanks. Well I think I'll just have to see how it goes...I dont record that much these days but anything I tape is generally loud. Plus I had some work gift cards so I was technically able to get the Pr-2 for free. :)
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Of all recorders currently available (other than using a phone to record along via a recording app), PR-2 is is least likely of any to set off a metal detector.
That said, metal detection will depend on how the sensitivity of the detector is set. I've seen some cranked up so high they detect as single small coin, even a small washer in a pocket. And others turned down so far they barely detect much of anything.
I just hold it above my head along with my phone and keys as they wand everything else down. No prob. If a walk-through you are paranoid about and don't trust, probably just put it under the phone, keys, wallet and any other stuff that goes in the basket. Its small and square and even with entry-level Jedi skills should be easy to pass off as a phone battery, pager, or remote of some kind if even noticed at all.
I agree with you there. I've gone through many walk throughs with my Edirol-R9 and I did not set it off. But youre right, they all have different sensitivities. So yeah most times you can hold it in your hand now with all your other stuff and no one bothers to look. The mics are a bigger deal to me, that has set off detectors in the past so I use a belt clip mounted glasses case that has a hidden pocket that fits the wires, putting everything into the basket now is not a problem and with the PR-2, that can go into my new wallet. Battery box is now a car key FOB so I just walk through as they pass everything to me.
pm coming
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It’s pretty common with the SP-CMC-8s and similar AT mics with the Low Sensitivity Mod that you’ll need to boost the gain a bit higher than usual—around 18 to 24 dB is normal with the PR-2. This helps prevent clipping and keeps your recordings clean. The good news is you can always bring the levels up further in post-production if needed. Hope that helps!
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
Thanks, ‘cause i fucked up my first recording with PR-2. I was between -12 db and -6 db…
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It’s pretty common with the SP-CMC-8s and similar AT mics with the Low Sensitivity Mod that you’ll need to boost the gain a bit higher than usual—around 18 to 24 dB is normal with the PR-2. This helps prevent clipping and keeps your recordings clean. The good news is you can always bring the levels up further in post-production if needed. Hope that helps!
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
Thanks, ‘cause i fucked up my first recording with PR-2. I was between -12 db and -6 db…
Another good point which I dont quite get but OK. All my prior recorders do not reflect that so basically keep it below -10 is the best route.
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It’s pretty common with the SP-CMC-8s and similar AT mics with the Low Sensitivity Mod that you’ll need to boost the gain a bit higher than usual—around 18 to 24 dB is normal with the PR-2. This helps prevent clipping and keeps your recordings clean. The good news is you can always bring the levels up further in post-production if needed. Hope that helps!
Also, remember that -10 db on the recorder actually corresponds to 0 dB in real life, so try not to raise it too high!
Thanks, ‘cause i fucked up my first recording with PR-2. I was between -12 db and -6 db…
Another good point which I dont quite get but OK. All my prior recorders do not reflect that so basically keep it below -10 is the best route.
Keep in mind that the suggestion to shoot for -10dB on the meters is only a very general guideline. -10dB on a VU meter doesn't necessarily correspond to a 0dB peak value.
The appropriate target value on the meters for "peaking sufficiently close to 0dBFS without going over" is going to vary with content and recording position, sometimes dramatically. That's because the crest factor (the ratio of fast transient peaks to steady-state slow-varying sound levels) of the music or other recorded content can vary significantly. A droning synth pad generally has a low crest-factor (the resulting VU and Peak meter levels will be similar), while a percussion hit will produce a high crest factor (Peak meter level will be high but VU meter level low). In terms of general concert taping, recording a PA amplified concert from the soundboard enclosure will mean a significantly lower crest factor (translating to somewhat higher values on the VU meters being OK) than when recording with mics placed on stage near the sound sources, particularly if close to the drum kit where there will be much higher transient peaks that will require targeting a significantly lower point on the VU meter to avoid peak overs.
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Keep in mind that the suggestion to shoot for -10dB on the meters is only a very general guideline. -10dB on a VU meter doesn't necessarily correspond to a 0dB peak value.
The appropriate target value on the meters for "peaking sufficiently close to 0dBFS without going over" is going to vary with content and recording position, sometimes dramatically. That's because the crest factor (the ratio of fast transient peaks to steady-state slow-varying sound levels) of the music or other recorded content can vary significantly. A droning synth pad generally has a low crest-factor (the resulting VU and Peak meter levels will be similar), while a percussion hit will produce a high crest factor (Peak meter level will be high but VU meter level low). In terms of general concert taping, recording a PA amplified concert from the soundboard enclosure will mean a significantly lower crest factor (translating to somewhat higher values on the VU meters being OK) than when recording with mics placed on stage near the sound sources, particularly if close to the drum kit where there will be much higher transient peaks that will require targeting a significantly lower point on the VU meter to avoid peak overs.
Thank you. That makes sense now. I've been doin a little more testing and feeling better about it. Got the USB to work too so I am happy about that but I still need to test out in the real world. I might have to hit this Heart concert for fun and see what happens. :)
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On-sale from Deity for $203 - https://shop.deitymic.com/products/pr-2
I just picked one up used from B&H...probably would have just done this instead, but whatever