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new member (field recordist)

jerryfreak

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Hi everybody,
I stumbled across your cool site today, and wanted to introduce myself

I do field recording with various portable microphone/preamp/ADC/recorder configurations and am a part of a larger group of same interest. Most people do recording of live performances, everything from 2 channel microphones at rock concerts to multiple microphone setups at classical halls/houses of worship etc. Most are amateurs as opposed to the guys who do audio recording on film sets and broadcast journalism, etc

We mostly hang out at our site taperssection.com (I'm same username over there). The scene grew out of the Grateful Dead taping scene in the 70s when people would lug large cassette decks and massive SLA batteries, and later the 2-piece PCM F1 setup. It has evolved over the years as a scene into itself with constantly evolving gear with the advent of all the new technologies

There is a different subset of field recording enthusiasts that do nature sounds, birding, etc. A little bit of overlap between the two groups as the gear is largely the same

a typical portable setup would include small diaphragm condenser microphones like schoeps/neumann/DPA/sennheiser/gefell/AT/AKG/etc. most of these require phantom power but there is always a move for more compact and efficient powering of microphones. In some cases, depending on the performance, small unobtrusive setups are required. For both reasons there has been a move towards everything from low-power mic preamp bodies intended for wireless transmitters (schoeps cmr, dpa mmp-g, etc), to custom-built miniature preamps, and other options, to drive the modular microphone capsules directly.

So there are a slew of different angles to that, often approached differently, such as variations on capsule polarization voltage, gain stage topology, etc. Custom solutions which come with of course varied opinions on their sound and specs.

Apart from the microphones themselves, proper powering (48V phantom) and preamplification was needed, and historically this was accomplished with portable preamps of old like the Grace Design V2, Sonosax SX-M2, Sound Devices MP2, etc. These would usually be followed by an ADC, anything from the sony SBM1, to the Apogee AD500 and AD1000, to the Benchmark AD2402-96

soon after (early 00s) we saw some nice integrated solutions like the Grace V3, Apogee Mini-me, and Sound Devices USB-pre. At the time 24-bit was the rage and we developed all kinds of ways to field record spdif into laptops and other handhelds like palm pilots etc.

late 00s brought some amazing all in one portable solutions like the Sound Devices 7xx series (722, 744,788, etc). these were expensive professional ENG solutions with timecode and a bunch of other things.

years later we are graced by many great affordable pro-sumer all-in-ones, from Roland, Zoom, Sound Devices among others

the latest trend is for recorders like the Sound Devices Mixpre II and Zoom F6 to use 2 or more ADCs at different levels and summing them to match the dynamic range of a high performance microphone input (~120-140 dBA EIN). These recorders are writing files to 32-bit float which removes the 144dB dynamic range limit of a 24-bit file, in favor a floating point (think of it as scientific notation for PCM data), which can account for an astonishing 1550 dB of dynamic range in the file.

Apart from the pro-sumer all-in-ones that generally have 2-8 or more channels of phantom power on balanced inputs, some of the other popular gear are good sounding handhelds (Sony, Roland, Tascam, etc) that take 1/8" in and sound good with a line-level signal from a portable preamp on an 1/8" minijack. This type of rig with a custom miniature mic pre, can drive a pair of SDCs and record a 24/96 file with adequate dynamic range, and fit in one pocket.

Long story short is we are a bunch of guys with a bunch of gear and perhaps we could collaborate a little to learn some things?
 
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restorer-john

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Long story short is we are a bunch of guys with a bunch of gear and perhaps we could collaborate a little to learn some things?

Awesome. Welcome!

Long into tape (cassette decks), open reel, even VHS HiFi back in the day, DAT (TC-D3 and TCD-7 and home/pro DATs) and a bit of poking about with a little Tascam of late recording birds and forest sounds around us. Those little handhelds are truly amazing for virtually no money.
 
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jerryfreak

jerryfreak

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yes they are! Here is a link to a site (commercial but hosts some good test data) that has all kinds of measurements on the handhelds

https://www.avisoft.com/recorder-tests/

one of the more popular ones in our world is the tiny Sony A10

most live shows that are recorded have dynamic range of 50-70 dB, so any of the handhelds approaching 90dB DR is adequate for that purpose


P.S. hope youre well with all the fires out there
 

SIY

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Hi everybody,
I stumbled across your cool site today, and wanted to introduce myself

I do field recording with various portable microphone/preamp/ADC/recorder configurations and am a part of a larger group of same interest. Most people do recording of live performances, everything from 2 channel microphones at rock concerts to multiple microphone setups at classical halls/houses of worship etc. Most are amateurs as opposed to the guys who do audio recording on film sets and broadcast journalism, etc

We mostly hang out at our site taperssection.com (I'm same username over there). The scene grew out of the Grateful Dead taping scene in the 70s when people would lug large cassette decks and massive SLA batteries, and later the 2-piece PCM F1 setup. It has evolved over the years as a scene into itself with constantly evolving gear with the advent of all the new technologies

There is a different subset of field recording enthusiasts that do nature sounds, birding, etc. A little bit of overlap between the two groups as the gear is largely the same

a typical portable setup would include small diaphragm condenser microphones like schoeps/neumann/DPA/sennheiser/gefell/AT/AKG/etc. most of these require phantom power but there is always a move for more compact and power efficient powering of microphones. In some cases, depending on the performance, small unobtrusive setups are required. For both reasons there has been a move towards everything from low-power preamps intended for wireless transmitters, to custom-built miniature preamps, and other options, to drive the modular microphone capsules directly.

So there are a slew of different angles to that, often approached differently, such as variations on capsule polarization voltage, gain stage topology, etc. Custom solutions which come with of course varied opinions on their sound and specs.

Apart from the microphones themselves, proper powering (48V phantom) and preamplification was needed, and historically this was accomplished with portable preamps of old like the Grace Design V2, Sonosax SX-M2, Sound Devices MP2, etc. These would usually be followed by an ADC, anything from the sony SBM1, to the Apogee AD500 and AD1000, to the Benchmark AD2402-96

soon after (early 00s) we saw some nice integrated solutions like the Grace V3, Apogee Mini-me, and Sound Devices USB-pre. At the time 24-bit was the rage and we developed all kinds of ways to field record spdif into laptops and other handhelds like palm pilots etc.

late 00s brought some amazing all in one portable solutions like the Sound Devices 7xx series (722, 744,788, etc). these were expensive professional ENG solutions with timecode and a bunch of other things.

years later we are graced by many great affordable pro-sumer all-in-ones, from Roland, Zoom, Sound Devices among others

the latest trend is for recorders like the Sound Devices Mixpre II and Zoom F6 to use 2 or more ADCs at different levels and summing them to match the dynamic range of a high performance microphone input (~120-140 dBA EIN). These recorders are writing files to 32-bit float which removes the 144dB dynamic range limit of a 24-bit file, in favor a floating point (think of it as scientific notation for PCM data), which can account for an astonishing 1550 dB of dynamic range in the file.

Apart from the pro-sumer all-in-ones that generally have 2-8 or more channels of phantom power on balanced inputs, some of the other popular gear are good sounding handhelds (Sony, Roland, Tascam, etc) that take 1/8" in and sound good with a line-level signal from a portable preamp on an 1/8" minijack. This type of rig with a custom miniature mic pre, can drive a pair of SDCs and record a 24/96 file with adequate dynamic range, and fit in one pocket.

Long story short is we are a bunch of guys with a bunch of gear and perhaps we could collaborate a little to learn some things?

Hi Jerry, it's something I love to do as well. Where are you?
 
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jerryfreak

jerryfreak

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Ivanovich

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Very cool, thanks for sharing!

I love me my playback, but I also dable in home studio stuff with guitar, bass, DIs, mics, and midi controller, etc. I build my hi-fi stereo speakers and mixing monitors, and absolutely love all the capabilities that new tech has now made available to enthusiasts.
 

NTomokawa

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Welcome!

Before I got into photography, I had myself a Sony PCM-D50 and would record sounds of traffic and nature for fun. Good times.
 
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jerryfreak

jerryfreak

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i personally found this thread very interesting

https://www.audiosciencereview.com/...-range-from-recording-to-listening-room.8205/

its topical as to what we do

for most of our needs, a low-sensitivity cardioid mic like schoeps MK4 or DPA 4011 with their self noise in the teens and max SPL around 130dB do the job.

That thread was eye opening as to the practical limits of psychoacoustics, where everybody is constantly "chasing the numbers". At the end of the day the practical constraints of your source material are the limiting factor and equipment capability (The old "id rather have the wrong mic in the right place than the right mic in the wrong place" philosophy).

The mic capsules above can be pushed to their absolute limit of performance with proper powering. For example the new schoeps CMC1 compact bodies improve over the classic CMC6 bodies by 4dB in max SPL. There are other custom mods (google Rens Heijnis or Sonodore) where the mic bodies are completely rebuilt to remove phantom power from the signal path to improve performance

in our art, where portability or unobtrusiveness is a factor, we often take shortcuts we know we can get away with, like deliberately underpowering microphone capsules to get similar sound at the expense of max SPL handling. some of these lower sensitivity options for driving capsules are the aforementioned schoeps CMR (https://schoeps.de/en/products/colette/microphone-amplifiers/cmr.html ) and DPA MMP-G (https://www.dpamicrophones.com/acce...ive-microdot-cable-for-pencil-microphones-3-m), both of which are compromises in a sense, but sip power - they both use ~1mA at 5V (compared for 3-4 mA @48V for their full sized counterparts). But I'm personally not close mic'ing a drum, so higher distortion above 120 dB could be irrelevant to me.

the latter DPA unit can utilize a clever preamp & AD powered by iphone lightning jack and achieve a claimed 114 dB dynamic range

https://www.dpamicrophones.com/accessories/mma-a-digital-audio-interface

modern designs have much better chips to work with, that tiny device sounds astonishingly good.

would love to get some measurements on the various different mic powering options, but im not sure anyone has access to the required equipment, something like this:

https://www.bksv.com/en/products/ca...systems/high-pressure-calibration-system-9719

there are some comparable legacy versions of the B&K equipment around as well, including the 4221 which is offered for rental: http://atecorp.com/ATECorp/media/pdfs/data-sheets/Bruel-Kjaer-4230_Brochure.pdf

i recall seeing briefly a poll here about testing speakers. microphones are kind of the same animal. you need to carefully account for the inherent distortion of the opposite transducer you are attempting to measure with. the specialized calibrated equipment almost seems essential

https://bksv.com/media/doc/083-80.pdf


back to the topic of this post in regard to the 120 dB thread, its also topical in the choice of ADCs employed in field recording. the newer multiple-ADC designs tout dynamic range in excess of 130 dB, which demonstrated by thread mentioned above, is above the practical limits of the quietest microphones used in the quietest studios played back on the the world's best playback setups. Thus the "need" for 32-bit FP recorders is a topic of discussion. In fact, some have debated these designs are disadvantageous relative to a standard HQ 24-bit ADC with its typical 115-120 dB dynamic range, as a result of the crossover distortion when switching between the multiple AD circuits, yet theres inherent practicality there when recording passages of unknown volume in unknown settings and want to come away with good results.

the "magic" of 32-bit float

https://www.sounddevices.com/noise-in-32-bit-float/

(theres also a bunch of youtube demos of the mixpre-II series and the Zoom F6 handling extreme variations in input levels if you care to look)
 
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gene_stl

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Welcome jerryfreak,

One of my favorite audiophile experiences was in the 1980s or so, a friend of mine here in St. Louis had me and another audio nut friend over.

He was a St. Louis symphony violinist and had just been to Japan to visit his family. He brought back a tiny Sony digital cassette recorder that was smaller than a microcassette. He handed it to each of us along with its earbuds. You listened and suddenly you were in Japan on his brothers patio with the breeze gently blowing and the birdies tweeting. It was a wonderful, memorable, and remarkable experience. I don't remember the model number but it certainly got added to my lust list.
 

scott wurcer

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i Thus the "need" for 32-bit FP recorders is a topic of discussion. In fact, some have debated these designs are disadvantageous relative to a standard HQ 24-bit ADC with its typical 115-120 dB dynamic range, as a result of the crossover distortion when switching between the multiple AD circuits, yet theres inherent practicality there when recording passages of unknown volume in unknown settings and want to come away with good results.

the "magic" of 32-bit float

32 Bit floats have no more DNR that 24 bit integers. The mantissa on IEEE 32bit floats is 24bits.
 
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jerryfreak

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scott wurcer

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its actually a 23 bit mantissa.... which gets multiplied by the exponent to represent a much larger range of data, and hence, dynamic range, of a digital audio signal... all of which are constrained to the very real limits of the analog input which precedes it.

Sorry sometimes miscount the sign bit. It's been a while looking at spectra converting 32bit floats to 24 vs 32 bit signed integers.
 

DonH56

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its actually a 23 bit mantissa.... which gets multiplied by the exponent to represent a much larger range of data, and hence, dynamic range, of a digital audio signal... all of which are constrained to the very real limits of the analog input which precedes it.

https://www.sounddevices.com/32-bit-float-files-explained/
https://en.wikipedia.org/wiki/IEEE_754

23 bits plus sign plus 8 exponent bits so effectively 24 bits (signed mantissa) plus exponent. Depends on whether you consider the sign part of the mantissa or not. I always have but apparently there are those who do not.

I used to do a fair amount of field recording using a fairly cheap Roland setup and/or little Zoom with some other mics. Have not done in years, however. Before that (long before) was live and studio stuff on R2R tape. Not so portable for the tape machines I had (always lusted after one of the "portable" R2R decks; I had several cheap smaller cassette and R2R decks but never an Ampex portable or the like, and my bigger units were a Studer console and back-up Pioneer RT-707 -- the latter I would haul on location now and then but it was only 7.5" and 7.5 ips vs. 15/30 ips and 10" reels on the console).
 
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jerryfreak

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im not really mathematically equipped to discuss it at that level

though it seems they are talking about precision of calculation of 32-bit float in DSP

i could kind of see this in a way. for a given mantissa value, there is a finite 'precision' those numbers can represent. and then the exponent in a way 'slides' that scale of representable numbers up and down, but it doesnt go any 'deeper' or more precise in calculations

seems like almost two different discussions. (DSP vs representing large differences in headroom)

im not super concerned about how the LSB in a DSP is handled, as much as i am about not clipping my signal by running out of data 'container capacity' for a better recording.

i did a test in soundforge. i generated a 1 sec 1khz sine at FSD and saved that file
then i used 'volume' effect in soundforge to adjust the volume by -120.00dB. saved that file
then i used volume again to adjust the volume +220.00 dB, which took it to an expected +100 dB signal on playback , saved that file
then i used volume to reduce it exactly -100.00 dB which put it at an expected 0dB peak

when i inverted this over the original 0dB generated file, it canceled it.... almost. there is some resulting random noise with peaks at -135 dB

so it can be argued that i indeed represented this waveform at ~24 bit precision at volume levels ranging over 200+ dB difference. however at the end of the day, the DSP precision of that file after multiple computations was indeed right around 24-bit.

could be multiple accumulated rounding errors at the 32-bit noise floor summing up to make some random low-level noise

regardless its impressive that i can even represent a waveform 99.9999999 something percent at hundreds of dB difference. considering how the ear works, there would never be a situation where you would need to represent the full range of dynamic range from loudest to softest that 32-bit floating point (or 24 bit fixed for that matter) can offer. From a practical perspective, being able to accurately recreate data thats even a few dB above FSD that would normally be atrociously clipped is a godsend.

and again, its no magic bullet. i still need to carefully match the output of my microphone to the input sensitivity of the recorder (+4dBU mic-in/+24dBU line-in), but the noise floor of the recorder itself is not a concern to me anymore, as it is below the level of the mic self-noise
 

scott wurcer

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im not really mathematically equipped to discuss it at that level

though it seems they are talking about precision of calculation of 32-bit float in DSP

i could kind of see this in a way. for a given mantissa value, there is a finite 'precision' those numbers can represent. and then the exponent in a way 'slides' that scale of representable numbers up and down, but it doesnt go any 'deeper' or more precise in calculations

seems like almost two different discussions. (DSP vs representing large differences in headroom)

Right, it's about what you could call instantaneous dynamic range. 24 bit signed integers effectively have 23 bits plus sign but any integer gain adjustment could either clip or throw away bits. Each operation in 32 bit floating point essentially normalizes the result to keep 23 bits plus sign in the result. There can be numerical noise issues when doing a large number of re-scales or effects but these are usually minor.
 

bennetng

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im not really mathematically equipped to discuss it at that level
This interactive demo should be self-explanatory:
https://www.h-schmidt.net/FloatConverter/IEEE754.html

An implied bit when exponent bits are non-zero (1), (23) bits of mantissa and (1) sign bit.

The benefit of floating point is it has a near constant precision at different scales, that makes digital processing and gain staging much safer and consistent.
 
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jerryfreak

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gotcha

so its a tradeoff. We want high precision (and 24 bit is more than adequate for all intents and purposes, 192 dB of representable dynamic range in 32 fixed point doesnt buy us much)

so in this case we are taking a very *practical* 24-ish bit representation of the signal and eliminating the catastrophic effects of running into artificailly imposed data limits with fixed point where 0 FSD= clipping

in this regard, for field recording purposes, wouldnt a simple 32-bit fixed signal, where you knew what your analog clipping input level was, with its 192 dB of range still be way overkill for representing the waveform. at MOST you might record something 100 dB down, the nuances of which would be cluttered by analog input noise anyway. from a practical perspective, i used to run schoeps CMRs thru a battery box and directly into a benchmark ad2402-96 at its most sensitive setting (+14dBU=0FSD). Poor match of the input levels but it was a minimalist 'the best preamp is no preamp' kind of mentality

heres an example of what it sounded like. outdoor venue (Red Rocks Amphitheater near Denver), where the crowd woulda been 40-50 dBA anyway.


all gain was done in post in digital, peaks of that recording were around -30 dBFS

seems like thats basically what these new 32-bit recorders are doing, matching an analog signal to a digital range and doing all work in the digital domain

the zoom F6 actually has no input level adjustment before the ADC stage. its a bit more sensitive ("native" is mic in at +4dBU max and they put a 20 dB pad inline for 'line' signals.)

technically if i ran the same low level signal of the CMRs thru the zoom's 'preamps' it would be similar to what i did with the benchmark (albeit with a signal more closely matched to the ideal dynamic range of the ADC circuit). any 'gain' applied would be post ADC
 
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