Become a Site Supporter and Never see Ads again!

Author Topic: What sample rate are you using?  (Read 23857 times)

0 Members and 1 Guest are viewing this topic.

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
What sample rate are you using?
« on: June 02, 2018, 09:55:02 PM »
I've been recording at 24/96 since I had a deck that could. Now I use Audacity in my post workflow, and it converts to 32/96, leading to large files. Hard drives are cheap, and I use an SSD as a work drive, so it's not a big problem, but I'm wondering what I'd lose going to 24/48 for recording.

Thoughts?

What rate do you use?

Thanks,
Chris
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline Fatah Ruark (aka MIKE B)

  • Trade Count: (11)
  • Needs to get out more...
  • *****
  • Posts: 9940
  • Gender: Male
  • I dream in beige.
    • sloppy.art.ink
Re: What sample rate are you using?
« Reply #1 on: June 02, 2018, 09:56:44 PM »
24/48.

For concerts I can't tell the difference above that.
||| MICS:  Beyer CK930 | DPA 4022 | DPA 4080 | Nevaton MCE400 | Sennheiser Ambeo Headset |||
||| PREAMPS: DPA d:vice | Naiant Tinybox | Naiant IPA |||
||| DECKS: Sound Devices MixPre6 | iPod Touch 32GB |||
|||Concert History || LMA Recordings || Live YouTube |||

Offline rocksuitcase

  • Trade Count: (4)
  • Needs to get out more...
  • *****
  • Posts: 8281
  • Gender: Male
    • RockSuitcase: stage photography
Re: What sample rate are you using?
« Reply #2 on: June 02, 2018, 10:13:57 PM »
24/48 live. Storage used to be the reason, now its doing 4 or 6 tracks at a time with the space/storage that entails.
music IS love

When you get confused, listen to the music play!

Mics:         AKG460|CK61|CK1|CK3|CK8|Beyer M 201E|DPA 4060 SK
Recorders:Marantz PMD661 OADE Concert mod; Tascam DR680 MKI x2; Sony PCM-M10

Offline heathen

  • Trade Count: (23)
  • Needs to get out more...
  • *****
  • Posts: 3528
Re: What sample rate are you using?
« Reply #3 on: June 02, 2018, 10:16:01 PM »
24/48. For the stuff I record, there's no reason for anything higher than that.
Mics: AT4050ST | AT4031 | AT853 (C/SC) | Line Audio CM3 | Sennheiser e614 | Sennheiser MKE2 | DPA 4061 Pre: CA9200 Decks: Zoom F8 | Roland R-05

Offline rigpimp

  • Trade Count: (14)
  • Needs to get out more...
  • *****
  • Posts: 3177
  • Gender: Male
  • Jarts don't kill people!
Re: What sample rate are you using?
« Reply #4 on: June 03, 2018, 01:04:19 AM »
24/96.  I have a 64TB NAS so space isn't an issue.
Mics: Schoeps MK 5 MP, Schoeps MK 8 MP, Schoeps MK 41 MP, KCY 250/5 > PFA
Pre/A>D/P48: Sonosax SX/M2, Sonosax SX/M2-LS, E.A.A. PSP-2, Baby Nbox, Neumann BS48i-2 (for sale)
Recorders: Sound Devices Mixpre-6ii, Sony PCM-A10
Playback: Jolida 1501 Hybrid > McIntosh MX 130 > Von Schweikert VR-4 JR, or Little Dot MK III > Sennheiser HD700
http://archive.org/bookmarks/kskreider
https://www.concertarchives.org/kskreider
https://archive.org/details/thespps

Offline firemt66

  • Trade Count: (2)
  • Taperssection Member
  • ***
  • Posts: 275
  • Gender: Male
Re: What sample rate are you using?
« Reply #5 on: June 03, 2018, 08:24:22 AM »
What is the sense of older shows recorded in 16bit re master to 96. The music has been recorded your not adding music..noise is being added,no?

Offline rippleish20

  • Trade Count: (24)
  • Taperssection All-Star
  • ****
  • Posts: 1021
Re: What sample rate are you using?
« Reply #6 on: June 03, 2018, 08:42:50 AM »
24/48
AKG C480B (ck61, ck63, ck8) /  Neumann KM100 (AK40, AK50) / AT853s  (cardiod, omni) / CA-14 (cardiod) / CA-11 (Omni) / Mixpre-10t / Mixpre-6 / Roland R-07 / Zoom F-3
paypal: rippleish20@gmail.com

Offline kindms

  • Trade Count: (6)
  • Needs to get out more...
  • *****
  • Posts: 5950
    • The Breakfast
Re: What sample rate are you using?
« Reply #7 on: June 03, 2018, 08:54:40 AM »
I've been recording at 24/96 since I had a deck that could. Now I use Audacity in my post workflow, and it converts to 32/96, leading to large files. Hard drives are cheap, and I use an SSD as a work drive, so it's not a big problem, but I'm wondering what I'd lose going to 24/48 for recording.

Thoughts?

What rate do you use?

Thanks,
Chris


audacity defaults to 32bit floating but you can change that in the interface after the import. Also you have your raw master, then your edited exported audacity at ? 24/96 or 16/44.1 why do you need to keep the 32 ?


my workflow is 24/48 original > Audacity edit in 32 floating > export to 24/48 and export to 16/44.1 > CD WAV Editor for tracking > TLH FLAC. I flac the untouched master files, the 24/48 tracked version for sharing and the 16/44.1 for sharing as well. Master files get archived, the tracked versions as well.  No need to keep the project files really just my .02
AKG c426, AKG414 XLS/ST, AKG ck61, ck22, >nBob colettes >PFA > V3, SD MixPre >  TCM-Mod Tascam HDP2, Sony M10
Little Bear tube Pre >Outlaw Audio 2200 Monoblocks > VR-2's

Offline wforwumbo

  • Trade Count: (11)
  • Taperssection Regular
  • **
  • Posts: 186
Re: What sample rate are you using?
« Reply #8 on: June 03, 2018, 09:01:28 AM »
I personally record at 24/96, thinking about moving up to 192. I don’t inherently hear a difference on the raw tapes, but many digital algorithms for things like EQs and saturation modeling plugins have significantly reduced filter error the higher the same rate; and that I do hear. Space is dirt cheap, I stash my shows on an external and have a 64-gig internal card in my mixpre-6.

What is the sense of older shows recorded in 16bit re master to 96. The music has been recorded your not adding music..noise is being added,no?

Assuming you mean 16 to 24 bit and/or 44.1k/48k/88.2k to 96k, correct. Depending on how the conversion is done there are ways to try and reduce noise but inherently it will add some level of noise. There are reasons why some may upsample, and some of them are valid (e.g. hardware is optimized for a certain bit depth/sample rate, or certain approximations in filter design improve drastically in phase response), but those are extreme special use cases, and far and few between.
North Jersey native, Upstate veteran, proud Texan

2x Schoeps mk2; 2x Schoeps mk21; 2x Schoeps mk4

4x Schoeps cmc5; 4x Schoeps KC5; Nbob KCY; Naiant PFA

EAA PSP-2

Sound Devices Mixpre-6

Offline Gordon

  • Trade Count: (22)
  • Needs to get out more...
  • *****
  • Posts: 11780
  • Gender: Male
    • my list
Re: What sample rate are you using?
« Reply #9 on: June 03, 2018, 10:01:19 AM »
24/48.

For concerts I can't tell the difference above that.
Microtech Gefell M20 or M21 > Nbob actives > Naiant PFA > Sound Devices MixPre-6 II @ 32/48

https://archive.org/details/fav-gordonlw

https://archive.org/details/teamdirtysouth

ilduclo

  • Guest
  • Trade Count: (0)
Re: What sample rate are you using?
« Reply #10 on: June 03, 2018, 10:40:21 AM »
24/44.1

Offline Sloan Simpson

  • Trade Count: (1)
  • Needs to get out more...
  • *****
  • Posts: 4013
  • Gender: Male
    • Southern Shelter
Re: What sample rate are you using?
« Reply #11 on: June 03, 2018, 10:59:54 AM »
24/48, and would do 44.1 except to head off complaints from video folks I send files to on occasion. I use Acustica plugins and processing is slow enough as it is at 48k; 96k rendering time is prohibitive.

ilduclo

  • Guest
  • Trade Count: (0)
Re: What sample rate are you using?
« Reply #12 on: June 03, 2018, 11:03:50 AM »
I need 24 bit to do post process amplification without adding hiss. But 44.1 works fine for sound quality, IMO and makes the files smaller and easier to work with.

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #13 on: June 03, 2018, 11:07:29 AM »
24/44.1

yep, but 24/48 if the video folks want some.

Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline billydee

  • Trade Count: (19)
  • Taperssection All-Star
  • ****
  • Posts: 1579
  • Gender: Male
Re: What sample rate are you using?
« Reply #14 on: June 03, 2018, 11:12:38 AM »
I guess I'll be the odd man out and say I only do 16/44 for 95% of what I record these days.

Offline u2_fly_2

  • Trade Count: (0)
  • Taperssection All-Star
  • ****
  • Posts: 1158
Re: What sample rate are you using?
« Reply #15 on: June 03, 2018, 11:33:37 AM »
96/24 Bit and also 192/24 on a new recorder. It also has the sound option "DSD Format with 2.8 MHz" which might or might not be to any use in the future(?)
Soundprofessionals Audio Technica AT 943 (SP-CMC-8) External Stereo Microphones > SP-SPSB-10-80020
Sound Professionals Micro-mini microphone power supply with mini 12vdc battery and 24 inch hardwired output cable Soundprofessionals Batterybox >> Olympus Ls-10 Linear Recorder > 4 GB > 24 Bit / 48 Khz  > 24 Bit / 96 Khz

Zoom Q3 HD - 1080p / 96-24 Bit

Roland R-26 (96 / 24 Bit)

Sony PCM-D100 (192 / 24 Bit)

Offline Scooter123

  • "I am not an alcoholic. I am a drunk. Drunks don't go to meetings."
  • Trade Count: (9)
  • Needs to get out more...
  • *****
  • Posts: 3774
Re: What sample rate are you using?
« Reply #16 on: June 03, 2018, 12:13:30 PM »
24 48

Regards,
Scooter123

mk41 > N Box  > Sony M-10
mk4 > N Box > Sony M-10

Offline jagraham

  • Trade Count: (20)
  • Needs to get out more...
  • *****
  • Posts: 2996
  • Gender: Male
Re: What sample rate are you using?
« Reply #17 on: June 03, 2018, 12:35:38 PM »
24/48 for the recording, mastering and archiving, dithered down to 16/44.1 for flac filesets for listening purposes. Of course I always keep the edited 24bit files along with the cue files so I could track them out in 24 later.
Mics: Nak CM-300s, Nak CM-100s, CP-1s, CP-2s, AT-853s(Cards, Hypers, Omnis) CA-14s(Cards, Omnis)
Pres: CA STC-9200, CA-UBB
Recorders: Tascam DR-70D, DR-2D, Edirol R-09

ISO: 1 Teac ME-120, CP-3 Caps, AT-853 Subcard Caps

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #18 on: June 03, 2018, 12:57:32 PM »
I guess I'll be the odd man out and say I only do 16/44 for 95% of what I record these days.




I record in 24bit to save my bacon when the levels get too hot, and I'm not watching..


I output 16/44.1 .flac when I'm finished post-processing, 100% of the time.




Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
Re: What sample rate are you using?
« Reply #19 on: June 03, 2018, 01:04:01 PM »
Seems like most use 24/48. I'll likely go to 48 khz instead of 96khz, as I haven't heard any argument against.

I think I'll stay with Audacity's 32 bit depth for post. From the manual:

"Audacity uses "float" format for 32-bit recording instead of fixed integer format as normalized floating point values are quicker and easier to process on computers than fixed integer values and allow greater dynamic range to be retained even after editing. This is because intermediate signals during audio processing can have very variable values. If they all get truncated to a fixed integer format, you can't boost them back up to full scale without losing resolution (i.e. without the data becoming less representative of the original than it was before). With floating point, rounding errors during intermediate processing are negligible.
The (theoretically audible) advantage of this is that 32-bit floating point format retains the original noise floor, and does not add noise. For example, with fixed integer data, applying a compressor effect to lower the peaks by 9 dB and separately amplifying back up would cost 9 dB (or more than 2 bits) of signal to noise ratio (SNR). If done with floating point data, the SNR of the peaks remains as good as before (except that the quiet passages are 9 dB louder and so 9 dB noisier due to the noise they had in the first place).
In many cases you will be exporting to a 16-bit format (for example if you are burning to a standard audio CD, that format is by definition 16-bit 44100 Hz). The advantage of using 32-bit float to work with holds even if you have to export to a 16-bit format. Using Dither on the Quality tab of Audacity Preferences will improve the sound quality of the exported file so there are only minimal (probably non-audible) effects of downsampling from 32-bit to 16-bit."

Note the use of 'theoretically' and 'probably'.
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #20 on: June 03, 2018, 01:18:03 PM »
Seems like most use 24/48. I'll likely go to 48 khz instead of 96khz, as I haven't heard any argument against.

I think I'll stay with Audacity's 32 bit depth for post. From the manual:

"Audacity uses "float" format for 32-bit recording instead of fixed integer format as normalized floating point values are quicker and easier to process on computers than fixed integer values and allow greater dynamic range to be retained even after editing. This is because intermediate signals during audio processing can have very variable values. If they all get truncated to a fixed integer format, you can't boost them back up to full scale without losing resolution (i.e. without the data becoming less representative of the original than it was before). With floating point, rounding errors during intermediate processing are negligible.
The (theoretically audible) advantage of this is that 32-bit floating point format retains the original noise floor, and does not add noise. For example, with fixed integer data, applying a compressor effect to lower the peaks by 9 dB and separately amplifying back up would cost 9 dB (or more than 2 bits) of signal to noise ratio (SNR). If done with floating point data, the SNR of the peaks remains as good as before (except that the quiet passages are 9 dB louder and so 9 dB noisier due to the noise they had in the first place).
In many cases you will be exporting to a 16-bit format (for example if you are burning to a standard audio CD, that format is by definition 16-bit 44100 Hz). The advantage of using 32-bit float to work with holds even if you have to export to a 16-bit format. Using Dither on the Quality tab of Audacity Preferences will improve the sound quality of the exported file so there are only minimal (probably non-audible) effects of downsampling from 32-bit to 16-bit."

Note the use of 'theoretically' and 'probably'.



Stick with Audacity 2.1.0 if you have to use the "Change Speed" tool.  Newer versions are broken, truncating error.  I have not examined V2.2.2, maybe someone can report on it.  V2.1.0 works perfectly, and is OK enough for me.






Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
Re: What sample rate are you using?
« Reply #21 on: June 03, 2018, 05:12:03 PM »
^ Thanks!
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline daspyknows

  • Complaint Dept.
  • Trade Count: (1)
  • Needs to get out more...
  • *****
  • Posts: 9636
  • Gender: Male
  • Don't ask, don't tell, don't get get caught
Re: What sample rate are you using?
« Reply #22 on: June 03, 2018, 08:15:39 PM »
24/48 for the recording, mastering and archiving, dithered down to 16/44.1 for flac filesets for listening purposes. Of course I always keep the edited 24bit files along with the cue files so I could track them out in 24 later.

same here

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #23 on: June 03, 2018, 09:47:04 PM »
I use 24/48, and share those and 16/44.1 “mastered” versions.
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline nulldogmas

  • Trade Count: (6)
  • Taperssection All-Star
  • ****
  • Posts: 1626
    • How I Escaped My Uncertain Fate
Re: What sample rate are you using?
« Reply #24 on: June 03, 2018, 10:16:41 PM »
24/48 for the recording, mastering and archiving, dithered down to 16/44.1 for flac filesets for listening purposes. Of course I always keep the edited 24bit files along with the cue files so I could track them out in 24 later.

same here

Yup, that.

Offline opsopcopolis

  • Trade Count: (3)
  • Needs to get out more...
  • *****
  • Posts: 2144
Re: What sample rate are you using?
« Reply #25 on: June 03, 2018, 11:53:57 PM »
16/44.1 usually. Occasionally 24/48. I almost never use 96 or higher even in my studio work (only really orchestral stuff/entirely acoustic music) so I sure as hell ain’t using it to record two track tapes of PA mixes from the back of a room.

Offline ycoop

  • Trade Count: (9)
  • Taperssection Member
  • ***
  • Posts: 647
  • Gender: Male
Re: What sample rate are you using?
« Reply #26 on: June 04, 2018, 03:21:53 AM »
I guess I'll be the odd man out and say I only do 16/44 for 95% of what I record these days.




I record in 24bit to save my bacon when the levels get too hot, and I'm not watching..


I output 16/44.1 .flac when I'm finished post-processing, 100% of the time.

This is something I've wondered about. What about the extra 8 bits allows for more work in post without losing as much quality?
Mics: Avantone CK-1s, AT853 c+o
Pres: CA9100
Recorders: DR-60d mkII, DR-2d

Offline aaronji

  • Site Supporter
  • Trade Count: (9)
  • Needs to get out more...
  • *
  • Posts: 3861
Re: What sample rate are you using?
« Reply #27 on: June 04, 2018, 06:48:43 AM »
This is something I've wondered about. What about the extra 8 bits allows for more work in post without losing as much quality?

A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise).  So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Offline yates7592

  • Trade Count: (12)
  • Taperssection Member
  • ***
  • Posts: 694
  • Gender: Male
Re: What sample rate are you using?
« Reply #28 on: June 04, 2018, 08:48:33 AM »
24/192 - storage space isn't an issue.

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #29 on: June 04, 2018, 12:52:08 PM »
This is something I've wondered about. What about the extra 8 bits allows for more work in post without losing as much quality?

A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise).  So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Precisely, thanks aaronji
Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #30 on: June 04, 2018, 05:56:20 PM »
The only thing that's important is that you're happy with your recordings.  I use 24/48, and I'm happy with that.  Depending on how you record, and your editing process, would make a difference as to what you should be recording at.  My suggestion, try out different things and determine what works best for you.
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Cheesecadet

  • Trade Count: (21)
  • Needs to get out more...
  • *****
  • Posts: 3016
  • Gender: Male
  • http://db.etree.org/cheesecadet
Re: What sample rate are you using?
« Reply #31 on: June 05, 2018, 01:02:01 AM »
24/44.1
AKG 481's, DPA 4061's (Matched), AT ES933's w/ AT853-ELE's (SC/C/O), ECM-19B's, MixPre-3, A10, M10 x 2, Hi Ho Silvers, Various Darktrain & GAKables

Vinyl:
Fluance RT83 Reference > MCS 3230 Receiver > Realistic MC-500's (NOS)

Upcoming:
02/10 J. Dilla
02/14 Wiliie Waldman & The Chocolate Factory
02/15 Gabe & Giz
02/16 GM5
02/28 Grace Potter

Offline morst

  • I think I found an error on the internet; #UnionStrong
  • Trade Count: (2)
  • Needs to get out more...
  • *****
  • Posts: 5950
Re: What sample rate are you using?
« Reply #32 on: June 05, 2018, 03:07:03 AM »
I record and work at 24 bits / 48 kHz from start to finish. Audacity is 32 bit internally, but I export as 24 bits.

Since I don't resample, I don't dither, but I would pick triangle dither if I had to do so.

I tried recording something at 96kHz once, and converted it down to 48kHz, then inverted and subtracted, and the difference file sounded blank to me. ¯\_(ツ)_/¯
https://toad.social/@morst spoutible.com/morst post.news/@acffhmorst

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #33 on: June 05, 2018, 09:59:55 AM »
I record at 24/48, edit in floating point (some plugins doing their own upsampling), then use triangular dither to truncate back to 24bit files for output.. or SRC to 44.1 + triangular dither to truncate to 16 bit.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline goodcooker

  • Trade Count: (43)
  • Needs to get out more...
  • *****
  • Posts: 4638
  • Gender: Male
  • goes to 11
Re: What sample rate are you using?
« Reply #34 on: June 05, 2018, 10:48:11 AM »
24/44.1

Same here. It's fine for 95% of what I do in the field.
Line Audio CM3/OM1 || MBHO KA500 hyper>PFA|| ADK A51 type IV || AKG C522XY
Oade Warm Mod and Presence+ Mod UA5s || Aerco MP2(needs help) || Neve Portico 5012 || Apogee MMP
SD Mixpre6 || Oade Concert Mod DR100mkii

pocket sized - CA11 cards > SP SB10 > Sony PCM A10

http://www.archive.org/bookmarks/goodcooker

"Are you the Zman?" - fan at Panic 10-08-10 Kansas City
"I don't know who left this perfectly good inflatable wook doll here, but if I'm blowing her up, I'm keeping her." -  hoppedup

Offline IMPigpen

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 255
Re: What sample rate are you using?
« Reply #35 on: June 05, 2018, 11:02:53 AM »
I use 24/48 for everything.  Most of my stuff I upload to the LMA.  I used to put up a 24/48 version and a 16/44 version but stopped doing that a while ago.  I've found that most people just listen to the streaming mp3 on there anyways.  And if archive my 24/48 raw files and the 24/48 FLAC.  So if someone wants something different, I can convert the FLACs to whatever they need.  I just had an artist ask me for mp3s of his solo show after I sent him the FLACs, so it took about 5 minutes to convert and zip them and send them to him.
Mics: Neumann KM184s; ADK TLs; Naiant X-R omnis & cards; MM-BSM-7M omnis
Pres/Battery boxes: Sound Devices MixPre; Naiant IPA; Sound Professionals SP-SPSB-6
Recorders: Tascam DR-70mkii; Tascam DR-60Dmkii; Tascam DR-40; Roland R-07
Cassette Transfers:  Nakamichi CR-1A->Tascam DR-60Dmkii
DAT Transfers:  Tascam DA-20mkii->Marantz PMD-670

My LMA uploads:  https://archive.org/details/@brandon_smith

**Have any old Strangefolk or Eddie From Ohio shows not on the LMA?  Let me know!**

Offline DSatz

  • Site Supporter
  • Trade Count: (35)
  • Needs to get out more...
  • *
  • Posts: 3347
  • Gender: Male
Re: What sample rate are you using?
« Reply #36 on: June 11, 2018, 09:09:07 PM »
I record at 24/44.1 if the eventual requirement is audio-only, and 24/48 if my sound track will be matched to video. Then I deliver whatever is specifically requested, whether it be mp3 files, CD-Rs, etc.

--aaronji wrote:

> A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise).  So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Aaronji, that just isn't true as a rule. It all depends on the noise in the signal that's being quantized versus the converter's own noise floor. The quantization noise of the converter only needs to be a few dB lower than the noise in the incoming analog signal, however many bits that takes. Any further bits beyond that will (or should!) be random; therefore they don't contribute to the precision or accuracy of the recording in any way. The notion that computation (post-processing) will somehow work out more accurately with more bits is a total fallacy when those bits contain only noise.

Of course if they contain meaningful signal information, then by all means keep them--but from what I've read on this board over the years, I think that a lot of people here are kind of fooling themselves on that point. For live classical recording, which is what I mostly do, the signal going into the a/d converter might have a noise floor maybe 60 or (if I'm very lucky) 65 dB below the peak levels. 14-bit quantization can handle that (keeping in mind that the converter noise needs to be below the incoming noise at all frequencies). 16-bit recording allows for more conservative level settings and fewer accidental overloads--plus it's been decades since 14-bit recording was even an option on available recorders. But for example the BBC used 10-bit (analog companded) studio-to-transmitter links for their classical broadcasts for years, and they were justly renowned for their audio quality.

24-bit recording (which might have 18 or 19 actual bits to offer, when the noise level of the converters and the rest of the signal chain is taken into account) is even nicer than 16, but it's clearly a luxury for our benefit as live engineers. I use it because it doesn't hurt anything*, and once in a blue moon it helps a little. But I've never delivered a 24-bit recording to a client; people not directly involved in recording have no use for the extra data, which only causes them trouble and expense.

(Notes: If the extra, unneeded converter bits beyond the incoming signal's noise floor aren't random, then they're adding distortion to the recording, and any such converter should be removed from service immediately. And please keep in mind that the noise floor of a converter is whatever its ACTUAL value is, with the number of bits setting a theoretical limit that is never reached in practice; no physically realizable 24-bit converter has an analog input noise floor anywhere near 144 dB below peak level. So please don't make a mental leap from "24 bits" to "8 bits quieter than a 16-bit converter" because it's much more likely in reality to be only 2 or 3 bits quieter, if that.)

--best regards

_________
* Edited later to add an asterisk above, and an important qualifying remark as a footnote: 24-bit recording "doesn't hurt anything" ONLY if you set your recording levels appropriately. At the risk of seeming to be unkind, let me say this as clearly as I can: If you're one of those people who completely misunderstand the concept of headroom, who actually AIM to achieve peak levels of -12 dB or even lower (instead of aiming to get your peak levels as close as possible to 0 dB without actually hitting it), then the use of 24-bit recording is hurting you by "enabling" your basic failure to understand the concept of recording levels, and your "24-bit" recordings are quite possibly noisier than they would be if you learned how digital recording actually works, and used 16-bit recording appropriately.
« Last Edit: June 11, 2018, 09:38:51 PM by DSatz »
music > microphones > a recorder of some sort

Offline morst

  • I think I found an error on the internet; #UnionStrong
  • Trade Count: (2)
  • Needs to get out more...
  • *****
  • Posts: 5950
Re: What sample rate are you using?
« Reply #37 on: June 12, 2018, 06:21:44 AM »
24-bit recording (which might have 18 or 19 actual bits to offer, when the noise level of the converters and the rest of the signal chain is taken into account) is even nicer than 16, but it's clearly a luxury for our benefit as live engineers. I use it because it doesn't hurt anything*, and once in a blue moon it helps a little.
I'll take it!
Quote
But I've never delivered a 24-bit recording to a client; people not directly involved in recording have no use for the extra data, which only causes them trouble and expense.
D'oh!
Quote
...So please don't make a mental leap from "24 bits" to "8 bits quieter than a 16-bit converter" because it's much more likely in reality to be only 2 or 3 bits quieter, if that.)

--best regards

_________
* Edited later to add an asterisk above, and an important qualifying remark as a footnote: 24-bit recording "doesn't hurt anything" ONLY if you set your recording levels appropriately. At the risk of seeming to be unkind, let me say this as clearly as I can: If you're one of those people who completely misunderstand the concept of headroom, who actually AIM to achieve peak levels of -12 dB or even lower (instead of aiming to get your peak levels as close as possible to 0 dB without actually hitting it), then the use of 24-bit recording is hurting you by "enabling" your basic failure to understand the concept of recording levels, and your "24-bit" recordings are quite possibly noisier than they would be if you learned how digital recording actually works, and used 16-bit recording appropriately.

Thanks for another informative post DSatz! Just for max clarity, could you please elaborate on the * part?
What would be the best practice for making use of 24-bit recording? Get as close as possible to 0 dBFS without going over, and don't just aim for a conservative -12 dB peak?
I must admit, I'm not scared of the occasional red light flashing on a transient peak, and I'm happy to find out if I can hear my limiters in action. (mostly never!)
https://toad.social/@morst spoutible.com/morst post.news/@acffhmorst

Offline heathen

  • Trade Count: (23)
  • Needs to get out more...
  • *****
  • Posts: 3528
Re: What sample rate are you using?
« Reply #38 on: June 12, 2018, 09:26:59 AM »
Just for max clarity, could you please elaborate on the * part?

Seconded.
Mics: AT4050ST | AT4031 | AT853 (C/SC) | Line Audio CM3 | Sennheiser e614 | Sennheiser MKE2 | DPA 4061 Pre: CA9200 Decks: Zoom F8 | Roland R-05

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #39 on: June 12, 2018, 10:59:21 AM »
What would be the best practice for making use of 24-bit recording? Get as close as possible to 0 dBFS without going over, and don't just aim for a conservative -12 dB peak?
I must admit, I'm not scared of the occasional red light flashing on a transient peak, and I'm happy to find out if I can hear my limiters in action. (mostly never!)

Get as close as possible to 0 dBFS without going over
^This is the optimal strategy for all digital recording.. if only it were so simple in the real world, where we don't know in advance how loud it will get.

The common misconception DSatz is referring to is the notion that peaking no higher than -12dB or whatever is somehow beneficial in and of itself.  Its not, its only good because it's not over. The goal isn't peaking specifically at -12dBfs, -6, -18, -3 or whatever number, the goal is simply not going over 0dBfs.  The second (unstated) goal which this achieves at the same time is to try and keep levels high enough that the quietest parts do not drop beneath the recording system's noise-floor.  That second part happens automatically if we push levels to get at close as possible to 0 dBfs, leaving almost all the "extra space" at the bottom as randomized noise.  A portion of that extra stuff at the bottom gets tossed out when truncating the 24 bit file down to 16 bits, but not all of it.  It's like trimming excess fat off that side, but still leaving a sufficeint fatty edge without cutting into the meaty portion. The full dynamics of the recording fits entirely within the 16bit file container.  It just has less excess fat than before.

Find a comfortable balance between not going over at the top of the dynamic range nor dropping down beneath the noise floor of the recording system at the bottom of the range.
^This strategy is effectively the same with respect to the end result, as long as the second half of the statement is true.  And it's a lot easier to manage in the real world.

It leaves more space at the top and somewhat less at the bottom as the recording is being made.  We can then trim some of that extra stuff from both the top and the bottom.  We cut excess fat from both sides this time, and end up with the same full dynamic recording in the 16 bit file.

If you never have a problem with noise-floor in the quiet parts of your recordings, then there is no problem with peaking somewhat lower and amplifying later.  It makes things easier in a practical sense by diminishing the possibility of overs.. or of limiters audibly cutting in.  Clipping doesn't sound good, and neither do limiters if you can audibly hear them working.  But otherwise there is no advantage to peaking lower other than not having to worry about overs.

No need to bump up too close to either end of the range. 17 or 18 bits worth of real-world recorded dynamic range leaves plenty of room in which to fit the full range of recorded sound with "headroom" and "footroom" to spare at both the top and bottom.

Sufficient headroom and footroom are not ends in themselves.  It's fitting comfortably within the limits which is the real goal.

musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline DSatz

  • Site Supporter
  • Trade Count: (35)
  • Needs to get out more...
  • *
  • Posts: 3347
  • Gender: Male
Re: What sample rate are you using?
« Reply #40 on: June 12, 2018, 11:07:04 PM »
Hmm. I'm not sure what needs to be explained, because I don't know what you currently know and/or believe. How about this as an outline:
  • Any audio channel--whether you're using it for recording or just to pass a signal along from point "A" to point "B"--has some overload limit and some noise "floor." The difference or distance between those two levels is that channel's dynamic range.
  • Meanwhile, any audio signal that comes from acoustic reality will also have some maximum (peak) level and some noise floor of its own; thus it has a dynamic range as well.
  • We want to record (or transmit, or pass along) our audio signal such that its full, audible dynamic range is preserved.
  • As long as the channel's dynamic range exceeds the signal's dynamic range by at least a handful of dB--at all frequencies and at all times throughout the duration of the program--then that channel can preserve that signal's dynamic range without audible degradation. You'll need to set the level relationship between the signal and the channel pretty much as Gutbucket describes (don't let your overs go over, or your unders go under)--and then the signal coming out of the channel will retain, for all audible purposes, the dynamic range that it had on its way in.
Thus as long as the channel has sufficient dynamic range and the levels are set optimally--where the peaks of the program material come as close as possible to the channel's maximum level without actually reaching it--any additional extension of the channel's dynamic range can't audibly improve things, because the signal itself is the limiting factor at that point, not the channel.

This is actually nothing peculiar to digital audio; analog works exactly the same way. In fact I would strongly recommend "thinking in analog" in order to get the correct mental picture.
« Last Edit: June 13, 2018, 02:23:46 AM by DSatz »
music > microphones > a recorder of some sort

Offline heathen

  • Trade Count: (23)
  • Needs to get out more...
  • *****
  • Posts: 3528
Re: What sample rate are you using?
« Reply #41 on: June 12, 2018, 11:27:15 PM »
Maybe I misunderstood the initial point you were making.  After reading the more recent posts, I'm assuming the initial point was that we shouldn't be using -12 (or -6 or -16, etc) dogmatically as our goal for where levels peak, but instead use 0.  But, based on Gutbucket's comment and your last post, if we don't get the levels right at 0 and fall a little under, it's okay as long as the low parts of the audio signal still ride above the noise floor of the recorder.  Is that (somewhat) accurate?

Sorry for my ignorance here...and thanks for everyone's patience and explanations.
Mics: AT4050ST | AT4031 | AT853 (C/SC) | Line Audio CM3 | Sennheiser e614 | Sennheiser MKE2 | DPA 4061 Pre: CA9200 Decks: Zoom F8 | Roland R-05

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
Re: What sample rate are you using?
« Reply #42 on: June 13, 2018, 12:26:45 AM »
Super appreciate where this thread has gone.

How does this relate to post-processing bit depth? Audacity defaults to 32 bit, any harm in going with 24?

TIA
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline edtyre

  • Trade Count: (85)
  • Needs to get out more...
  • *****
  • Posts: 2261
  • Gender: Male
  • Team Philly " No Excuses, Just Tapes"
Re: What sample rate are you using?
« Reply #43 on: June 13, 2018, 02:01:31 AM »
24/44.1 or 24/48 no need for anything else
Tryed DSD, 24/192 doesn’t work for me.
music>mics>pre>recorder

Offline DSatz

  • Site Supporter
  • Trade Count: (35)
  • Needs to get out more...
  • *
  • Posts: 3347
  • Gender: Male
Re: What sample rate are you using?
« Reply #44 on: June 13, 2018, 02:06:57 AM »
Heathen, that's exactly right as far as the general relationship between a signal and a channel is concerned.

However, a recorder's circuit design also matters. When the record level control on a recorder is set way too low, there's a real risk that unnecessary, extra noise is being added to the recording that doesn't come only from the noise floor of the a/d converter stage, but also from the noise floor of one or more earlier stages of the analog electronics. The user is basically inviting extra, unnecessary noise into the recording at each analog stage after the record level control.

One well-known recorder that I'm almost certain has this characteristic, for example, is the Sony PCM-M10. According to what I saw when I did some testing and measuring about five years ago, a 16-bit recording that reaches -2 on peaks would be considerably quieter than an exactly equivalent 24-bit recording that reaches only -12 but is then boosted 10 dB in post-processing. That's the opposite of the result that I think some people would expect.

I should probably repeat those measurements to make sure, but this isn't so much about that particular recorder as it is about the fact that there are always multiple analog stages before the a/d converters in any recorder--and if you set the record levels too low (or the sensitivity switch to its less sensitive setting when the more sensitive setting could be used safely without overload), it is rather likely that there will be unnecessary, audible noise contributions from one or more stages of the electronics prior to the converter.

Ideally, for the best signal-to-noise ratio. you want to nearly "saturate" each successive analog stage before the converters--keep them pumped full of signal, so that their noise doesn't make an audible contribution. Again, that's a basic way of thinking that an analog engineer would have from experience, that people have lost sight of in the digital era. I think some people might think about individual sample values too much, and/or think about them as if they only contain energy from the desired signal. But any one quantized sample reflects the instantaneous sum of the desired signal plus noise--and for any given sample, you can never know its "value due to signal" or its "value due to noise," just as you can't "unstir" a cup of coffee that already has cream and sugar in it.

--best regards
« Last Edit: June 13, 2018, 12:40:06 PM by DSatz »
music > microphones > a recorder of some sort

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #45 on: June 13, 2018, 09:16:55 AM »
Dsatz is one of the main reasons that Ts.com remains the best place on the internet.

Thanks for your continued presence here.
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline aaronji

  • Site Supporter
  • Trade Count: (9)
  • Needs to get out more...
  • *
  • Posts: 3861
Re: What sample rate are you using?
« Reply #46 on: June 13, 2018, 10:21:06 AM »
--aaronji wrote:

> A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise).  So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Aaronji, that just isn't true as a rule. It all depends on the noise in the signal that's being quantized versus the converter's own noise floor. The quantization noise of the converter only needs to be a few dB lower than the noise in the incoming analog signal, however many bits that takes. Any further bits beyond that will (or should!) be random; therefore they don't contribute to the precision or accuracy of the recording in any way. The notion that computation (post-processing) will somehow work out more accurately with more bits is a total fallacy when those bits contain only noise.

I have to admit (and with all due respect) that I am not quite sure what you mean.  My comment had nothing to do with post-processing accuracy.  What I was saying is that, as I understand it, the major (only?) advantage of the extra bits is to increase the channel's dynamic range by reducing quantization noise.  That makes it easier to fit the dynamic range of the signal, even with very low peak levels, into that space with (to use Gutbucket's term) sufficient footroom.  So when amplified, the signal's noise will dominate the converter's noise (rendering it inaudible).  Am I missing something here?  There are even some examples available (such as on the Sound Device's website) that appear to show this and it also seems to be in line with both your post and Gutbucket's.

Granted, the real world, live music recordings scenarios in which this will actually help are likely rare, but stuff happens due to accident or circumstance...

Offline H₂O

  • Trade Count: (28)
  • Needs to get out more...
  • *****
  • Posts: 5745
  • Gender: Male
Re: What sample rate are you using?
« Reply #47 on: June 13, 2018, 10:50:19 AM »
1bit 2.8224Mhz for 2 channel
24bit 96Khz for 4-8 channel
Music can at the least least explain you and at the most expand you
LMA Recordings

List

Offline rigpimp

  • Trade Count: (14)
  • Needs to get out more...
  • *****
  • Posts: 3177
  • Gender: Male
  • Jarts don't kill people!
Re: What sample rate are you using?
« Reply #48 on: June 13, 2018, 11:26:08 AM »
Dsatz is one of the main reasons that Ts.com remains the best place on the internet.

Thanks for your continued presence here.

This^^.  If I am perusing and I see a post of his I take off my coat and read his post thoroughly.  I do the same with a few others like Morst and Gutbucket, for example.  The historical and technical contributions these folks make is so valuable. 

With that said, I am still going to run at 24/96.  :-p
Mics: Schoeps MK 5 MP, Schoeps MK 8 MP, Schoeps MK 41 MP, KCY 250/5 > PFA
Pre/A>D/P48: Sonosax SX/M2, Sonosax SX/M2-LS, E.A.A. PSP-2, Baby Nbox, Neumann BS48i-2 (for sale)
Recorders: Sound Devices Mixpre-6ii, Sony PCM-A10
Playback: Jolida 1501 Hybrid > McIntosh MX 130 > Von Schweikert VR-4 JR, or Little Dot MK III > Sennheiser HD700
http://archive.org/bookmarks/kskreider
https://www.concertarchives.org/kskreider
https://archive.org/details/thespps

Offline wforwumbo

  • Trade Count: (11)
  • Taperssection Regular
  • **
  • Posts: 186
Re: What sample rate are you using?
« Reply #49 on: June 13, 2018, 02:53:17 PM »
I agree with much of what’s said here, particularly coming from DSatz (whose praises I want to echo throughout). BUT... I DO want to iron out something here, and I’ve brought it up before. The “1 bit == 6 dB of dynamic range” statement.

That is a theoretical rule that mostly follows how bits work at lower bit depths. A 24-bit word length will always contain 24 bits of information. It’s not that the extra bits are wasted; they are still there, and they are accessible in memory and computation. Where we run into limitations is in the design of ICs (mostly in amplifier design and topology) and getting THOSE low-noise enough. That is a pretty complex and technical challenge.

Something else worth noting... the average dynamic range of a young, healthy human hearing system is about 110 dB of dynamic range. So as long as you are peaking below 0 dB-FS, the purpose of having more bits and a wider dynamic range is to make it such that your ears - not the technical limitations of your system - is the bottleneck for getting things to sound “as good as you can” so to speak.

And again, another advantage of larger word lengths, is reduced filter error. The more data you are pumping through a system doing truncated (discrete) calculations, the more accurate the end product will be. It’s why 32-bit floating point operations are used internally in DAWs, and we aren’t stuck to using just 16- or 24-bit systems. Post work does improve significantly in my (and many other studio pros’) experience and to my/our ears.

Again, this isn’t to say that practically most of what’s said here is right, and it’s not to say that you can get away with 16/44.1 without issue; in many cases, it will do the job just fine. Just want to clarify, from a technical perspective.
North Jersey native, Upstate veteran, proud Texan

2x Schoeps mk2; 2x Schoeps mk21; 2x Schoeps mk4

4x Schoeps cmc5; 4x Schoeps KC5; Nbob KCY; Naiant PFA

EAA PSP-2

Sound Devices Mixpre-6

Offline goodcooker

  • Trade Count: (43)
  • Needs to get out more...
  • *****
  • Posts: 4638
  • Gender: Male
  • goes to 11
Re: What sample rate are you using?
« Reply #50 on: June 13, 2018, 03:30:41 PM »
the average dynamic range of a young, healthy human hearing system

I hope you realize that a lot of us here are old, worn out, half deaf and barely human... :guitarist: :zombie02:
Line Audio CM3/OM1 || MBHO KA500 hyper>PFA|| ADK A51 type IV || AKG C522XY
Oade Warm Mod and Presence+ Mod UA5s || Aerco MP2(needs help) || Neve Portico 5012 || Apogee MMP
SD Mixpre6 || Oade Concert Mod DR100mkii

pocket sized - CA11 cards > SP SB10 > Sony PCM A10

http://www.archive.org/bookmarks/goodcooker

"Are you the Zman?" - fan at Panic 10-08-10 Kansas City
"I don't know who left this perfectly good inflatable wook doll here, but if I'm blowing her up, I'm keeping her." -  hoppedup

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #51 on: June 13, 2018, 07:35:36 PM »
I'm happy wforwumbo has joined TS.  As far as I can tell by way of a few personal conversations he is sharp, extremely well versed in matthmatics, not old, only partially deafened thus far (yet retains a very keen ear), and is fully human.  Keep an eye open for his future contributions on the board.

I do want to bring up something which came to mind while reading his post above, because it is something I find commonly misunderstood at TS and could be easily misconstrued from what he wrote.  I regularly see statements in these kinds of threads which say something to this effect- "I record 24 bit files because I plan to do post work on the recordings, and 24 bit files are better for that", but that doesn't actually correlate with what he wrote above. 

Performing mathematics with sufficient precision such that rounding/truncation errors are avoided makes for more accurate calculations, and that can translate to better sounding audio.  But the bit-length of the data being manipulated doesn't dictate the bit-length of the calculation space.  The calculation space within the DAW is larger (32-bit floating point) so calculation precision is preserved.. until one outputs it again.  As DSatz describes above-

Quote
..as long as the channel has sufficient dynamic range and the levels are set optimally--where the peaks of the program material come as close as possible to the channel's maximum level without actually reaching it--any additional extension of the channel's dynamic range can't audibly improve things, because the signal itself is the limiting factor at that point, not the channel.

The raw 16 or 24 bit recorded file is a channel containing the signal.  Mathematics are performed on that signal in the DAW in a 32-bit floating point space regardless of the signal's bit-length.  We then export the data from the DAW via another file (channel) of sufficient bit-length to contain the manipulated signal.  Because we have full control over signal levels when manipulating and exporting that signal from the DAW (unlike when we were originally making the raw recording) we can make sure the signal fits comfortably within an output channel of an optimum size, so we can actually output a 16 bit file for almost everything without compromising the signal. We can make sure that "the signal itself is the limiting factor at that point, not the channel".  Or we can output 24 bit files with some extra unrelated noise at the bottom, which is easy, and space is cheap.  But in most cases a 16 bit output file can be of equivalent quality because the full dynamic range of almost all music can be fully represented by less than 16 bits.

The portions quoted below should be understood as being relevant within the DAW (or possibly within the recorder prior to a file being written in some cases), and not as specifying what file length is necessary in file storage formats-

A 24-bit word length will always contain 24 bits of information. It’s not that the extra bits are wasted; they are still there, and they are accessible in memory and computation.

Quote
The more data you are pumping through a system doing truncated (discrete) calculations, the more accurate the end product will be. It’s why 32-bit floating point operations are used internally in DAWs, and we aren’t stuck to using just 16- or 24-bit systems. Post work does improve significantly in my (and many other studio pros’) experience and to my/our ears.



tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

Sample rate conversion is a different animal.
« Last Edit: June 13, 2018, 07:39:35 PM by Gutbucket »
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
Re: What sample rate are you using?
« Reply #52 on: June 13, 2018, 08:35:11 PM »
....
tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

For math-challenged folks, what's the tldr of the tldr?

I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

Thanks for simplifying way way down.
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline wforwumbo

  • Trade Count: (11)
  • Taperssection Regular
  • **
  • Posts: 186
Re: What sample rate are you using?
« Reply #53 on: June 13, 2018, 11:46:50 PM »
....
tldr- As long as the full dynamics of the raw recording fits within 16 bits, recording at 24 bits will not make "post processing more accurate or sound better", but doing the post processing in a larger bit-length calculation space can do so. 

For math-challenged folks, what's the tldr of the tldr?

I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

Thanks for simplifying way way down.

Try this post I wrote:

http://taperssection.com/index.php?topic=184569.msg2251598#msg2251598

Still on the technical side, but a simple explanation of the overall concept.
North Jersey native, Upstate veteran, proud Texan

2x Schoeps mk2; 2x Schoeps mk21; 2x Schoeps mk4

4x Schoeps cmc5; 4x Schoeps KC5; Nbob KCY; Naiant PFA

EAA PSP-2

Sound Devices Mixpre-6

Offline checht

  • Site Supporter
  • Trade Count: (5)
  • Taperssection Member
  • *
  • Posts: 765
  • Let's meet at alternate foods at the break
Re: What sample rate are you using?
« Reply #54 on: June 14, 2018, 12:52:41 AM »
^ That clarifies a lot.

Thanks!
Schoeps MK41 x 2, MK22 x 2; Vanguard V1s matched pair; Niaint x8
Schoeps kcy5, nbob actives
Naiant PFA 60v, PFA 48v, IPA
Sound Devices MP-6II; Sony PCM-A10

Recordings at LMA

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #55 on: June 14, 2018, 10:50:43 AM »
Quote
I record trying to get close to but not exceed 0db. Will my recordings be better in 16 or 24 bit?

In practical terms it likely depends on what recorder you are using, the ADC chip and specific implementation of it in the machine.  If you don't have measurements, don't care to do critical listening tests, don't want to worry about it too much, and don't care about storage size, my take is that it makes sense to default to 24 bit recording in this day and age, if simply as a belt and suspenders approach more than a definitely better kind of thing. This is a practical take rather than an engineering answer.


Practicality rules.  In the end, I suspect the two largest practical determinants on this is what era one started recording in, and how many TB of raw recordings one has amassed and needs to deal with! Storage space may be growing ever cheaper, but managing and backing up large catalogs is a PITA.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline DSatz

  • Site Supporter
  • Trade Count: (35)
  • Needs to get out more...
  • *
  • Posts: 3347
  • Gender: Male
Re: What sample rate are you using?
« Reply #56 on: June 14, 2018, 11:40:34 AM »
aaronji, howdy. As for why I mentioned post-processing in my reply to you, please look again at the earlier message of yours that I replied to. It quotes a message from ycoop as follows: "What about the extra 8 bits allows for more work in post without losing as much quality?"--to which you replied that essentially, more bits in a/d conversion = lower quantization error. To which, in turn, I replied that no, that's not a valid general statement; it all depends on the noise in the signal being converted.

If that noise isn't correlated with the desired signal, and is entirely above the noise floor of the converter, then more bits don't (can't!) improve the accuracy of the conversion. Hypothetically and simplistically, if the noise floor of a signal is at a 13-bit level, you could convert that signal with a 14-bit, 16-bit, 20-bit or 24-bit converter all of which would give equal audible results (all other things being equal), because the noisy signal was the limiting factor, not the converter.

However, if the noise IS correlated with the desired signal, then it is a form of non-linear distortion and you've got yourself a defective converter. Most discussion assumes tacitly that the residual noise of a converter will be uncorrelated with the signal, because for decades now, that has normally been the case. That is another way of saying that the non-linear distortion is very low.

Still, the older ones among us, or anyone who's ever used an inadequately dithered A/D converter, will know what that kind of distortion sounds like, because some of the earliest available digital audio recording equipment didn't dither properly (ahem Sony, including their professional PCM-1600 which a lot of early CDs were recorded with). Some people describe it as "noise breathing" or "noise pumping" because it rises and falls with the signal levels, particularly on very soft sounds such as reverberation decay (or imagine taking a 400 Hz tone and fading it down slowly to nothingness with a smooth, analog fader). An accompanying phenomenon is (or was) "digital deafness"--signals below the scope of the lowest-order bit are simply blanked out.

Unfortunately, the way a lot of people think about digital audio is still based on the incredible flood of bullshit that was unleashed when digital audio was introduced around 1980, and which (when it was about anything real) harped greatly on shortcomings of conversion that were really due to not having proper dither. When an a/d converter is properly dithered, there's no increase in distortion at lower levels, no "stair-step levels" to be minimized by striving for more and more bits, nor is there "digital deafness" at the bottom of the range; it acts just like an analog channel as far as dynamic range is concerned. Since actual distortion in modern a/d converters is so low, I tend to shift the discussion away from the concept of "precision" in conversion (which would be a more appropriate concept if a signal were utterly noiseless, such as a computationally generated test signal) to dynamic range, which is much more pertinent to live, recorded audio.

--best regards
« Last Edit: June 14, 2018, 11:53:32 AM by DSatz »
music > microphones > a recorder of some sort

Offline wforwumbo

  • Trade Count: (11)
  • Taperssection Regular
  • **
  • Posts: 186
Re: What sample rate are you using?
« Reply #57 on: June 14, 2018, 02:30:36 PM »
aaronji, howdy. As for why I mentioned post-processing in my reply to you, please look again at the earlier message of yours that I replied to. It quotes a message from ycoop as follows: "What about the extra 8 bits allows for more work in post without losing as much quality?"--to which you replied that essentially, more bits in a/d conversion = lower quantization error. To which, in turn, I replied that no, that's not a valid general statement; it all depends on the noise in the signal being converted.

If that noise isn't correlated with the desired signal, and is entirely above the noise floor of the converter, then more bits don't (can't!) improve the accuracy of the conversion. Hypothetically and simplistically, if the noise floor of a signal is at a 13-bit level, you could convert that signal with a 14-bit, 16-bit, 20-bit or 24-bit converter all of which would give equal audible results (all other things being equal), because the noisy signal was the limiting factor, not the converter.

However, if the noise IS correlated with the desired signal, then it is a form of non-linear distortion and you've got yourself a defective converter. Most discussion assumes tacitly that the residual noise of a converter will be uncorrelated with the signal, because for decades now, that has normally been the case. That is another way of saying that the non-linear distortion is very low.

Still, the older ones among us, or anyone who's ever used an inadequately dithered A/D converter, will know what that kind of distortion sounds like, because some of the earliest available digital audio recording equipment didn't dither properly (ahem Sony, including their professional PCM-1600 which a lot of early CDs were recorded with). Some people describe it as "noise breathing" or "noise pumping" because it rises and falls with the signal levels, particularly on very soft sounds such as reverberation decay (or imagine taking a 400 Hz tone and fading it down slowly to nothingness with a smooth, analog fader). An accompanying phenomenon is (or was) "digital deafness"--signals below the scope of the lowest-order bit are simply blanked out.

Unfortunately, the way a lot of people think about digital audio is still based on the incredible flood of bullshit that was unleashed when digital audio was introduced around 1980, and which (when it was about anything real) harped greatly on shortcomings of conversion that were really due to not having proper dither. When an a/d converter is properly dithered, there's no increase in distortion at lower levels, no "stair-step levels" to be minimized by striving for more and more bits, nor is there "digital deafness" at the bottom of the range; it acts just like an analog channel as far as dynamic range is concerned. Since actual distortion in modern a/d converters is so low, I tend to shift the discussion away from the concept of "precision" in conversion (which would be a more appropriate concept if a signal were utterly noiseless, such as a computationally generated test signal) to dynamic range, which is much more pertinent to live, recorded audio.

--best regards

Well-stated, eloquently and accurately. I 100% agree with every aspect of this post.

And while the digital “crunch” you mention from poor converters is considered charming or nostalgic by some, it definitely isn’t pleasant to listen to. I’ve heard my fair share of bad converters and understand why digital gets such a bad reputation. 

DSatz I hope I’m blessed enough to tape with you at some point and learn a thing or two from your experience.
North Jersey native, Upstate veteran, proud Texan

2x Schoeps mk2; 2x Schoeps mk21; 2x Schoeps mk4

4x Schoeps cmc5; 4x Schoeps KC5; Nbob KCY; Naiant PFA

EAA PSP-2

Sound Devices Mixpre-6

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #58 on: June 14, 2018, 02:44:47 PM »
Wow, a big thank you to DSatz, and as always Gutbucket as well as others which made this stuff so much easier to understand.  Luckily, I've been doing the right thing, more or less, possibly, as pointed out above, because I'm an old timer that started out with analogue. Reading these specs put everything in perspective and helped me to understand why I do what I do, and that I may even be able to improve my craft a little to boot.  Can't wait to experiment...Thank you!
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline robeti

  • Trade Count: (39)
  • Taperssection Member
  • ***
  • Posts: 904
Re: What sample rate are you using?
« Reply #59 on: June 14, 2018, 04:01:17 PM »
I tape, archive and listen at 24/48

mics schoeps mk22/mk4/mk41 (matched) | nakamichi cm-300 (JB mod/cp1/cp2/cp3) | nakamichi cm-50 | primo em4052pmi4's | sp-cmc-4u/at-853 4.7k mod (shotguns/h/c/sc/o) | ca-11 c/o | ca-14 c
power ca-ubb | ca-9200 | nbob actives > baby nbox | schoeps cmbi (pair)
recorder roland r-05 
video panasonic zs100 | panasonic hdc-sd600 | sony hx9v | sony hx50v | samsung s23 ultra
playback fiio m17 > final d8000

Offline aaronji

  • Site Supporter
  • Trade Count: (9)
  • Needs to get out more...
  • *
  • Posts: 3861
Re: What sample rate are you using?
« Reply #60 on: June 14, 2018, 04:49:36 PM »
aaronji, howdy.

Howdy yourself, DSatz!  Thank you for the detailed response!

Since actual distortion in modern a/d converters is so low, I tend to shift the discussion away from the concept of "precision" in conversion (which would be a more appropriate concept if a signal were utterly noiseless, such as a computationally generated test signal) to dynamic range, which is much more pertinent to live, recorded audio.

To be honest, I have viewed bit depth solely in terms of dynamic range, based on a lot of articles I have read.  I guess I am one of those folks that just assumed decent conversion.  I always figured that 24-bit would give an extra cushion if I left way too much headroom (as has happened by accident or circumstance a few times), and I will still use it when recording, but the likelihood that the noise floor of anything I record even approaches 16 bits is virtually nil...

Offline sparko

  • Trade Count: (3)
  • Taperssection Newbie
  • *
  • Posts: 36
  • Gender: Female
Re: What sample rate are you using?
« Reply #61 on: June 25, 2018, 05:43:30 AM »
Since I never heard a difference between 96 and 48, I now record mostly in 24/48, my recording app sometimes switches itself to 32/48, producing larger files. (no difference to hear as well) I keep forgetting to check the setting before I hit record.
I work with Audition and I do all post process in 16/44.1; EQ'ing, cutting, flac conversion and sharing

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #62 on: June 25, 2018, 08:56:09 AM »
Since I never heard a difference between 96 and 48, I now record mostly in 24/48, my recording app sometimes switches itself to 32/48, producing larger files. (no difference to hear as well) I keep forgetting to check the setting before I hit record.
I work with Audition and I do all post process in 16/44.1; EQ'ing, cutting, flac conversion and sharing

You really should be post-processing those 24 bit files, THEN dithering to 16 bits when you finish.

Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline sparko

  • Trade Count: (3)
  • Taperssection Newbie
  • *
  • Posts: 36
  • Gender: Female
Re: What sample rate are you using?
« Reply #63 on: June 25, 2018, 10:36:46 AM »
tbh I never heard a difference in the mixing, neither 24/96, 24/48 or 16/44.1, on any of my speakers, devices, headphones whatsoever.

Offline beenjammin

  • Trade Count: (1)
  • Taperssection Regular
  • **
  • Posts: 107
Re: What sample rate are you using?
« Reply #64 on: June 25, 2018, 11:08:07 AM »

(Notes: If the extra, unneeded converter bits beyond the incoming signal's noise floor aren't random, then they're adding distortion to the recording, and any such converter should be removed from service immediately. And please keep in mind that the noise floor of a converter is whatever its ACTUAL value is, with the number of bits setting a theoretical limit that is never reached in practice; no physically realizable 24-bit converter has an
* Edited later to add an asterisk above, and an important qualifying remark as a footnote: 24-bit recording "doesn't hurt anything" ONLY if you set your recording levels appropriately. At the risk of seeming to be unkind, let me say this as clearly as I can: If you're one of those people who completely misunderstand the concept of headroom, who actually AIM to achieve peak levels of -12 dB or even lower (instead of aiming to get your peak levels as close as possible to 0 dB without actually hitting it), then the use of 24-bit recording is hurting you by "enabling" your basic failure to understand the concept of recording levels, and your "24-bit" recordings are quite possibly noisier than they would be if you learned how digital recording actually works, and used 16-bit recording appropriately.

DSatz, another great post! I have a question that I hope is not off topic. (I'm a nature recordist and diehard AUD tape/acoustic music listener and so hide out here in hopes of learning something from this great forum).

What about those of use who don't aim at -12dB (or lower) but who don't really have a choice given the source material?

I regularly record nature sounds which are frequently at such low levels. Save for some wind gusts or a jet engine, the usable recording I have to work with hovers down there. I opt not to attain "unnatural" levels when working in post, but I do routinely add as much as +10 dB so that the recording is audible at appropriate levels.

Should I experiment with 16bit?

Offline morst

  • I think I found an error on the internet; #UnionStrong
  • Trade Count: (2)
  • Needs to get out more...
  • *****
  • Posts: 5950
Re: What sample rate are you using?
« Reply #65 on: June 25, 2018, 06:36:12 PM »
What about those of use who don't aim at -12dB (or lower) but who don't really have a choice given the source material?

I regularly record nature sounds which are frequently at such low levels. Save for some wind gusts or a jet engine, the usable recording I have to work with hovers down there. I opt not to attain "unnatural" levels when working in post, but I do routinely add as much as +10 dB so that the recording is audible at appropriate levels.

If wind gusts and jet engine sounds make those portions of the recording unusable, then why even consider them when setting levels? If they'll be thrown away, who cares how much they are clipped? Set the levels for the sounds you want.

JUST BE CAUTIOUS WHEN MONITORING THIS STUFF!
https://toad.social/@morst spoutible.com/morst post.news/@acffhmorst

Offline beenjammin

  • Trade Count: (1)
  • Taperssection Regular
  • **
  • Posts: 107
Re: What sample rate are you using?
« Reply #66 on: June 25, 2018, 08:41:30 PM »
What about those of use who don't aim at -12dB (or lower) but who don't really have a choice given the source material?

I regularly record nature sounds which are frequently at such low levels. Save for some wind gusts or a jet engine, the usable recording I have to work with hovers down there. I opt not to attain "unnatural" levels when working in post, but I do routinely add as much as +10 dB so that the recording is audible at appropriate levels.

If wind gusts and jet engine sounds make those portions of the recording unusable, then why even consider them when setting levels? If they'll be thrown away, who cares how much they are clipped? Set the levels for the sounds you want.

JUST BE CAUTIOUS WHEN MONITORING THIS STUFF!

Yes, fully agree! I guess I wasn't clear: I have my levels cranked to 11 and am still lower than -12dB. My question is: given DSatz insights about 16 vs 24 bit, perhaps I should be recording at 16bit?

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #67 on: June 25, 2018, 09:26:50 PM »
tbh I never heard a difference in the mixing, neither 24/96, 24/48 or 16/44.1, on any of my speakers, devices, headphones whatsoever.

That may be true, but others might have ears or gear which allow for hearing a difference, and, more importantly, if you ever want to do an post processing, many plugins work better at a higher sample rate.
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline morst

  • I think I found an error on the internet; #UnionStrong
  • Trade Count: (2)
  • Needs to get out more...
  • *****
  • Posts: 5950
Re: What sample rate are you using?
« Reply #68 on: June 26, 2018, 01:35:39 AM »
Yes, fully agree! I guess I wasn't clear: I have my levels cranked to 11 and am still lower than -12dB. My question is: given DSatz insights about 16 vs 24 bit, perhaps I should be recording at 16bit?
I'd be more interested in seeing what you can get if you could crank your levels to 23 and see what is there.

What if you got a microphone which was more sensitive, or a preamp with more (clean??!!) gain?
https://toad.social/@morst spoutible.com/morst post.news/@acffhmorst

Offline jerryfreak

  • No PZ
  • Trade Count: (31)
  • Needs to get out more...
  • *
  • Posts: 6205
  • The plural of anecdote is not data
Re: What sample rate are you using?
« Reply #69 on: June 26, 2018, 04:54:01 AM »
i usually record at 24/44.1

most of the recorders im using these days have not been fully tested by me for buffer underruns/dropped samples id rather reduce the load by 50+% than chase the holy grail of 96K and up.

older i get the safer i play it (as in life)

24 vs 16 bit is absolutely essential imo esp if youre taping different content where levels will vary dramatically.

with my AD2K i could get crystal clear sound in excess of my mics noise floor peaking at -30 dB and never need a preamp or worry about checking or setting levels. couldnt pull that off with 16bit
Unable to post or PM due to arbitrary censorship of people the mod doesn't like. Please email me using the link in my profile if you need to connect

Offline beenjammin

  • Trade Count: (1)
  • Taperssection Regular
  • **
  • Posts: 107
Re: What sample rate are you using?
« Reply #70 on: June 26, 2018, 11:01:21 AM »
Yes, fully agree! I guess I wasn't clear: I have my levels cranked to 11 and am still lower than -12dB. My question is: given DSatz insights about 16 vs 24 bit, perhaps I should be recording at 16bit?
I'd be more interested in seeing what you can get if you could crank your levels to 23 and see what is there.

What if you got a microphone which was more sensitive, or a preamp with more (clean??!!) gain?

Again I wasn't clear, sorry: by "11" I'm referring to the movie Spinal Tap and mean that all my levels, preamp and line are set to maximum. In light of what DSatz has just said about 24bit vs 16bit above and given  that some places I record are extremely quiet, where levels hover lower than -12dB with everything cranked up, I wonder if 16bit might actually make sense for me.

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #71 on: June 26, 2018, 12:14:42 PM »
^ No.  At least that's not the appropriate conclusion to be drawn from what he posted.

By his measurements of the Sony PCM-M10, a 16bit recording which peaks at -2dBfs captures more dynamic range (has a lower noise floor) than the same recording made at 24bit which peaks at -12dBfs and then has 10 dB of gain added afterwards to bring it up to the same -2dB peak level as the 16 bit recording.

If you are unable to apply sufficient gain to get your peak levels higher than -12dB, then switching to 16bit will not help (and is likely to hurt, but to be certain of that would require additional testing and measurement).

His point was to not leave excessive headroom by shooting for some arbitrarily lower peak level simply because you've set the recorder to write 24bit files.  Instead, the goal remains peaking as high as practical without going over regardless of the bit depth of the file being written, allowing for sufficient headroom to avoid overs, but no more than really needed.   He was making a point which applies generally, by way of measurements made of the Sony PCM-M10 specifically, regarding a characteristic which is likely to apply to other recorders in the same class commonly used by tapers.

The practical take away-
You don't have the option of using more gain and are stuck with peaking at -12dB regardless of how you set the recorder.  Lacking hard evidence to the contrary it's probably best for you to stick with 24 bits as you are likely to produce a recording with noise floor a couple of dB lower than you would if you were recording 16bits and also peaking at -12dB.   A couple dB lower is better, but it isn't a full 8 bits worth lower (the difference between 24 and 16).  That couple of dB reflects the likely real-world performance increase from 16 to 24 bit modes in small low cost recorders. 

This all translates to 24bit mode on commonplace recorders providing just a couple more dB of usable headroom than 16bit mode.  Not 10dB more or whatever as many would expect, but just a few dB more.  Use it but don't abuse it.

Here's the relevant portion of his post-
Quote
One well-known recorder that I'm almost certain has this characteristic, for example, is the Sony PCM-M10. According to what I saw when I did some testing and measuring about five years ago, a 16-bit recording that reaches -2 on peaks would be considerably quieter than an exactly equivalent 24-bit recording that reaches only -12 but is then boosted 10 dB in post-processing. That's the opposite of the result that I think some people would expect.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline beenjammin

  • Trade Count: (1)
  • Taperssection Regular
  • **
  • Posts: 107
Re: What sample rate are you using?
« Reply #72 on: June 26, 2018, 09:36:51 PM »
^ Thanks! T

hat's very helpful and makes a lot of sense. I had read DSatz post and realized how little I know about how digital recording actually works and started to question my entire approach to sampling and bit depth. I'll keep on keeping on, cranking the dial in the quietest of places.

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #73 on: June 28, 2018, 04:25:28 PM »
Gutbucket (or others)...

I'm interested in what you would do, with regards to levels, when recording 4 tracks (say, 2 from a soundboard and 2 from microphones...or 2 different pair of microphones).  Would you still "saturate" as much as possible without going over, and lower the levels in post before combining, or would you record a little lower to make post easier?
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #74 on: June 28, 2018, 06:42:38 PM »
Good question.  The previous, but in practical terms the answer becomes "both".

Generally its a good idea to set gain for each pair individually at the start of recording, then re-adjust as appropriate during mixing.

When mixing, I always balance levels and make correction tweaks by ear anyway.  That's a critical part of my post processing, even more so when mixing multiple channels, where I balance each pair separately before balancing the mix of pairs.  So in that sense I'm not overly concerned with the particular level relationship between multiple channels while recording since I'm going to adjust things by ear later.  From that perspective I'm setting gains so as to maximize dynamics for each channel individually while recording.

But..

Once the show gets rolling and I have the recording gains pretty much dialed in where I want them, at that point I don't want to make gain changes to individual channels because doing so would throw off whatever delicate mix balance I reach afterwards, from that particular point on in the performance.  So once I dial in whatever recording gains I feel are appropriate for each channel, I link the gain controls across all recording channels so that I can adjust them all simultaneously.  That way if I need to reduce levels for any one channel I adjust the gain of all of them together using one knob.  I hopefully don't need to do that during any one set, but often do between sets for multiple artist acts, such as at a festival.  Then whenever I do adjust gains during recording, the level change is obvious in post and I can locate and compensate for that for all channels across the board (adjusting them all by the same amount), without throwing off my mix balance.

Since I'm personally recording six or sometimes more channels, all of which are part of a multiple microphone array where each channel has a specific relationship to the others, including a tightly-correlated 3-microphone "center pair", doing this becomes even more important than it does when managing four channels.

It's sort of like the ambisonic case described above, except I don't need to precisely match identical recording gain across all channels.  I only need to keep the relative gain between channels the same throughout the recording.

Keeping the relative gains the same is a practicality of mixing thing.

I'll occasionally check the lowest level channels, typically rearward facing "ambience" channels, to make sure the signal remains above the noise floor of my recording system during the quietest parts.  If it does, no worries.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #75 on: June 28, 2018, 07:14:56 PM »
Good question.  The previous, but in practical terms the answer becomes "both".

Generally its a good idea to set gain for each pair individually at the start of recording, then re-adjust as appropriate during mixing.

When mixing, I always balance levels and make correction tweaks by ear anyway.  That's a critical part of my post processing, even more so when mixing multiple channels, where I balance each pair separately before balancing the mix of pairs.  So in that sense I'm not overly concerned with the particular level relationship between multiple channels while recording since I'm going to adjust things by ear later.  From that perspective I'm setting gains so as to maximize dynamics for each channel individually while recording.

But..

Once the show gets rolling and I have the recording gains pretty much dialed in where I want them, at that point I don't want to make gain changes to individual channels because doing so would throw off whatever delicate mix balance I reach afterwards, from that particular point on in the performance.  So once I dial in whatever recording gains I feel are appropriate for each channel, I link the gain controls across all recording channels so that I can adjust them all simultaneously.  That way if I need to reduce levels for any one channel I adjust the gain of all of them together using one knob.  I hopefully don't need to do that during any one set, but often do between sets for multiple artist acts, such as at a festival.  Then whenever I do adjust gains during recording, the level change is obvious in post and I can locate and compensate for that for all channels across the board (adjusting them all by the same amount), without throwing off my mix balance.

Since I'm personally recording six or sometimes more channels, all of which are part of a multiple microphone array where each channel has a specific relationship to the others, including a tightly-correlated 3-microphone "center pair", doing this becomes even more important than it does when managing four channels.

It's sort of like the ambisonic case described above, except I don't need to precisely match identical recording gain across all channels.  I only need to keep the relative gain between channels the same throughout the recording.

Keeping the relative gains the same is a practicality of mixing thing.

I'll occasionally check the lowest level channels, typically rearward facing "ambience" channels, to make sure the signal remains above the noise floor of my recording system during the quietest parts.  If it does, no worries.

OK, I completely agree with what you said...especially when running more that 4 mics, with a "center".  I often run 5 mics (or 3 mics with a board patch - still 5 channels), and the blend(s) mic(s) can really throw things off.

Since I'm not concerned with a balanced mix of all tracks, not being concerned with everything being equal when recording.  I will, more often than not, lean heavier in one direction, depending on the sound of the tracks.  (eg. 60/40...75/25...55/50...)

What I was really getting at though, is when recording each source, knowing that you'll have to lower your levels before blending the sources, do you tend to set your levels lower than if recording 2 tracks, or record as if they're separate sources?
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #76 on: June 29, 2018, 10:05:30 AM »
I record them as if separate sources.
Mix them by ear.
Adjust output level by eye.

In other words-
I record optimizing recording gains for each.  Import the files and adjust the relative levels of those sources in the mixing software by ear, listening for whatever sounds best and works right (sometimes going back and forth between a few different "good" choices before making a decision, but never by any specific ratio or dB difference), and adjust the resulting level in the mix bus by eye, checking the peak-hold meter so that the highest peak ends up around -.5 dBfs or so.

musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #77 on: June 29, 2018, 10:15:23 AM »
I'm always wonder a bit when folks state specific mix ratios between sources here at TS.  How do you know what the ratio is? I can't imagine folks are doing RMS ratio calculations to determine what the ratio might be. Is there some "combine" function of specific mixing software that specifies this? 

If so, unless the separate sources had exactly the same levels prior to mixing, any level difference between the files would be invisible to the mixing software (unless the software was pretty advanced).  It's basically the same situation as the Mid/Side ratio thing I described earlier in the thread - the Mid/Side controller assumes the Mid and Side channels were recorded with equal sensitivity microphones and recording gains, and if that's not the case the ratio displayed on the controller display is not the actual ratio.

A dB difference between sources would seem to me easier to determine and more useful (being a logarithmic rather than a linear relationship) if one wanted to describe the relationship between sources to someone else, just get a numeric "feel" for a specific level difference, or determine how much difference is in play before one source effectively renders the other inaudible.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #78 on: June 29, 2018, 11:15:50 AM »
I'm always wonder a bit when folks state specific mix ratios between sources here at TS.  How do you know what the ratio is? I can't imagine folks are doing RMS ratio calculations to determine what the ratio might be. Is there some "combine" function of specific mixing software that specifies this? 

If so, unless the separate sources had exactly the same levels prior to mixing, any level difference between the files would be invisible to the mixing software (unless the software was pretty advanced).  It's basically the same situation as the Mid/Side ratio thing I described earlier in the thread - the Mid/Side controller assumes the Mid and Side channels were recorded with equal sensitivity microphones and recording gains, and if that's not the case the ratio displayed on the controller display is not the actual ratio.

A dB difference between sources would seem to me easier to determine and more useful (being a logarithmic rather than a linear relationship) if one wanted to describe the relationship between sources to someone else, just get a numeric "feel" for a specific level difference, or determine how much difference is in play before one source effectively renders the other inaudible.

"I record them as if separate sources.
Mix them by ear.
Adjust output level by eye.

In other words-
I record optimizing recording gains for each.  Import the files and adjust the relative levels of those sources in the mixing software by ear, listening for whatever sounds best and works right (sometimes going back and forth between a few different "good" choices before making a decision, but never by any specific ratio or dB difference), and adjust the resulting level in the mix bus by eye, checking the peak-hold meter so that the highest peak ends up around -.5 dBfs or so."


OK, I am clearly not articulating myself well on this topic.  My apologies for all the confusion.

I essentially do the same thing you do...as stated above.

My original question was only relating to your original recording levels, which you did answer.  As you are aware, when recording multi tracks, and then combining them, they increase which can then create a handful of peaks.  If, when recording the live source, you lower your levels based on what you'd normally record at, then you'd be raising your levels in post if they are not already the way you like them once you mix tracks.  When recording as you normally would with two tracks, levels will have to be lowered before mixing tracks.  My question was based on that.  After getting the swing of multi-track recording, I lowered my original recording levels by about 3dB in anticipation of post production.  I've never had a problem with this (yet).  In other cases, when I want the different microphone sources to also shine on their own, I will record as in they are pairs only, and end up having to lower the levels in post.

The ratios a I sighted were bullshit and I shouldn't have stated them that way, I just thought for the sake of simplification, it would best describe what I was trying to say.  I was obviously incorrect.  My sources usually differ from being the same to a 3-4 dB difference (before mixing, in order to attain the sound I like best).  That is to say, If one pair of mics are Naks and the other pair are Gefells, when mixing, I will determine what I need more in the mix (by ear), and adjust accordingly.  If I remember correctly, my most recent 4-track recording ended up at a 2dB difference between the two source pair of mics.  Naks are warmer and provide a more resonating bottom end, while the Gefells offer a crisp, more precise high end.  They blend nicely, but it's a matter of preference what your ear may prefer.

I hope that helps clarify what I meant to say.

Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #79 on: June 29, 2018, 03:25:54 PM »
I lowered my original recording levels by about 3dB in anticipation of post production.  I've never had a problem with this (yet). 

Indeed, as you articulate, combining two (roughly identical) sources results in a 3db increase in level. That's because the decibel is a unit of measurement used to express the ratio of one value of a physical property to another on a logarithmic scale. So, when you reduce two inputs which will eventually be combined by 3db, you are compensating for this doubling. Or, to say it differently, if you combine two identical tracks, you increase the level by 3db.

However, I don't see any benefit whatsoever in reducing levels while recording, and advise you do it differently. You'd be much better off in trying to get as close to 0 without going over on every channel as you do when you run only two channels. Then, using a DAW, reduce the level of individual channels in post to taste when mixing. Start with each track at -3db and go from there, and put a brick wall limiter at -.01 db, as the last plugin.

Gutbucket (or others)...

I'm interested in what you would do, with regards to levels, when recording 4 tracks (say, 2 from a soundboard and 2 from microphones...or 2 different pair of microphones).  Would you still "saturate" as much as possible without going over, and lower the levels in post before combining, or would you record a little lower to make post easier?

I'd put it this way:
lower the levels in post while combining

« Last Edit: June 29, 2018, 03:30:08 PM by noahbickart »
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #80 on: June 29, 2018, 03:30:45 PM »
Was typing this as Noah posted, which basically states the same less concisely..

As you are aware, when recording multi tracks, and then combining them, they increase which can then create a handful of peaks.  If, when recording the live source, you lower your levels based on what you'd normally record at, then you'd be raising your levels in post if they are not already the way you like them once you mix tracks.  When recording as you normally would with two tracks, levels will have to be lowered before mixing tracks.  My question was based on that.  After getting the swing of multi-track recording, I lowered my original recording levels by about 3dB in anticipation of post production.  I've never had a problem with this (yet).  In other cases, when I want the different microphone sources to also shine on their own, I will record as in they are pairs only, and end up having to lower the levels in post.

There is no need to record the source files which are likely to be mixed together later at a lower level.  The editing software should provide multiple points of gain adjustment.  Like an analog mixing console which a digital multichannel editor emulates, one can adjust gain of the individual tracks/input channels as well as the output level from the mix bus. The mix relationship is controlled by the relative gains applied at the individual track/channel level, and the output level controlled by the gain of the mix bus.  One can also control the levels through the mix bus by collectively lowering the gain of all individual tracks/channels by equal amounts, but it's easier to do so with the mix bus output gain control.

A well designed analog mixing console provides sufficient summing headroom through the mix bus to accommodate the increase in level that arises from the combination of individual channels.  One needs to keep each input channel from clipping, and as long as that is the case there should be sufficient headroom internally to mix all channels together without clipping. Then the gain through the mix bus output amplifier can be adjusted so that it does not clip and whatever it is feeding doesn not clip.

In a digital multitrack editor, the internal calculations are typically done with 32 bit floating point mathematical precision. That provides "mathematical headroom" somewhat analogous to the voltage headroom inside a well designed analog console.  You need to get the signal into the editor without going over and out of the editor without going over, but within the editor itself the calculation space is large enough to provide sufficient headroom until dithered and truncated down to a lower bit depth output file
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #81 on: June 29, 2018, 04:16:35 PM »
combining two (roughly identical) sources results in a 3db increase in level.

This is generally true for mixing in practice, within a few constraints- First the two sources need to have close to the same level to begin with.  If one source has a significantly higher level than the other, the sum of the two will not have a significantly higher level than the higher level source.  For mix combinations where the levels of the two sources are not quite identical but close enough that the combination is musically meaningful (you can hear the contribution of the lower level source) the sum will be less.

Assuming the two sources are exactly same level prior to summing and have a positive phase correlation..

If the sources are incoherent the sum is +3dB
If the sources are coherent the sum is +6dB

Most sources we will be mixing together will be have a coherence somewhere in between those extremes yet closer to incoherent than coherent, so the sum will generally be much closer to +3dB than +6db.

Coherent means the signals completely identical in content and phase.
Incoherent means a randomized phase relationship between the signals. 

Good examples of primarily-incoherent signals are those from microphones spaced far enough apart, or directional microphones pointing far enough away from each other, or both.  An AUD and a SBD will be predominantly incoherent as the sources are sampled from different physical locations.  A good example of full coherence would be summing a source with a copy of itself.

If the sources have a negative phase correlation the resulting level of the sum is reduced not increased.  Consider the polarity of once source flipped with respect to the other.  In that case the sum of the same two incoherent sources will be -3dB instead of +3dB.  And the sum of two fully coherent signals, say a source and its polarity inverted copy, will be...   a mathematical null (-∞dB), but in the real world a quite low but not infinitely attenuated level.  Because mathmatics is absolute but the real world is fuzzy.

« Last Edit: June 29, 2018, 04:22:43 PM by Gutbucket »
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #82 on: June 29, 2018, 04:26:08 PM »
I lowered my original recording levels by about 3dB in anticipation of post production.  I've never had a problem with this (yet). 

Indeed, as you articulate, combining two (roughly identical) sources results in a 3db increase in level. That's because the decibel is a unit of measurement used to express the ratio of one value of a physical property to another on a logarithmic scale. So, when you reduce two inputs which will eventually be combined by 3db, you are compensating for this doubling. Or, to say it differently, if you combine two identical tracks, you increase the level by 3db.

However, I don't see any benefit whatsoever in reducing levels while recording, and advise you do it differently. You'd be much better off in trying to get as close to 0 without going over on every channel as you do when you run only two channels. Then, using a DAW, reduce the level of individual channels in post to taste when mixing. Start with each track at -3db and go from there, and put a brick wall limiter at -.01 db, as the last plugin.

Gutbucket (or others)...

I'm interested in what you would do, with regards to levels, when recording 4 tracks (say, 2 from a soundboard and 2 from microphones...or 2 different pair of microphones).  Would you still "saturate" as much as possible without going over, and lower the levels in post before combining, or would you record a little lower to make post easier?

I'd put it this way:
lower the levels in post while combining

Yes Noah, this is basically what I was thinking.  After running a couple of 4 tracks, I liked having the "individual pairs" nicely recorded.  Dropping them down a few dB in post is no big deal.  I was just interested in what others did...

At the Citi Field shows, I was FOB with Jon and took a digital line out of his 722/Gefells while also running my Naks.  As anticipated, they played very nicely together.  If I could ever figure out how to post a damn show on bt.etree, I would like to put them out there.
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #83 on: June 30, 2018, 10:08:14 PM »
I lowered my original recording levels by about 3dB in anticipation of post production.  I've never had a problem with this (yet). 

Indeed, as you articulate, combining two (roughly identical) sources results in a 3db increase in level. That's because the decibel is a unit of measurement used to express the ratio of one value of a physical property to another on a logarithmic scale. So, when you reduce two inputs which will eventually be combined by 3db, you are compensating for this doubling. Or, to say it differently, if you combine two identical tracks, you increase the level by 3db.

However, I don't see any benefit whatsoever in reducing levels while recording, and advise you do it differently. You'd be much better off in trying to get as close to 0 without going over on every channel as you do when you run only two channels. Then, using a DAW, reduce the level of individual channels in post to taste when mixing. Start with each track at -3db and go from there, and put a brick wall limiter at -.01 db, as the last plugin.

Gutbucket (or others)...

I'm interested in what you would do, with regards to levels, when recording 4 tracks (say, 2 from a soundboard and 2 from microphones...or 2 different pair of microphones).  Would you still "saturate" as much as possible without going over, and lower the levels in post before combining, or would you record a little lower to make post easier?

I'd put it this way:
lower the levels in post while combining

Yes Noah, this is basically what I was thinking.  After running a couple of 4 tracks, I liked having the "individual pairs" nicely recorded.  Dropping them down a few dB in post is no big deal.  I was just interested in what others did...

At the Citi Field shows, I was FOB with Jon and took a digital line out of his 722/Gefells while also running my Naks.  As anticipated, they played very nicely together.  If I could ever figure out how to post a damn show on bt.etree, I would like to put them out there.

I’d be happy to walk you through it at some point, get in touch.
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #84 on: July 02, 2018, 01:49:05 PM »
Quote from: noahbickart link=topic=186648.msg2269858#msg2269858 date=1530410894[/quote

I’d be happy to walk you through it at some point, get in touch.

LOL, my friend, I need more than a walk through!  I need a "Posting For Dummies" printout...  I sat in with my friend Joe (not sure if you know him, but he goes by Joe Beacon).  He went through it with me step by step, and I wrote it all down.  I think I was able to get it to work once.  I think.  I may have a different version than him for something, but all it does is stress me out.  If you feel like dealing with it, and want to write out a step by step (taking nothing for granted), I'm a creature of habit and would follow the directions verbatim, and start posting lots of goodies.  I would only post to bt.etree, just so you know.  When I want something posted, I give it to Joe, but don't always like to ask because he has enough of his own stuff to deal with, so mine often doesn't get posted.
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline noahbickart

  • phishrabbi
  • Site Supporter
  • Trade Count: (33)
  • Needs to get out more...
  • *
  • Posts: 2554
  • Gender: Male
  • So now I wander over grounds of light...
Re: What sample rate are you using?
« Reply #85 on: July 02, 2018, 03:31:50 PM »
Quote from: noahbickart link=topic=186648.msg2269858#msg2269858 date=1530410894[/quote

I’d be happy to walk you through it at some point, get in touch.

LOL, my friend, I need more than a walk through!  I need a "Posting For Dummies" printout...  I sat in with my friend Joe (not sure if you know him, but he goes by Joe Beacon).  He went through it with me step by step, and I wrote it all down.  I think I was able to get it to work once.  I think.  I may have a different version than him for something, but all it does is stress me out.  If you feel like dealing with it, and want to write out a step by step (taking nothing for granted), I'm a creature of habit and would follow the directions verbatim, and start posting lots of goodies.  I would only post to bt.etree, just so you know.  When I want something posted, I give it to Joe, but don't always like to ask because he has enough of his own stuff to deal with, so mine often doesn't get posted.

Well, I teach people how to understand the Talmud for a living, and I guarantee that bt.etree.org is less obtuse than the Babylonian Talmud....
Recording:
Capsules: Schoeps mk41v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, GAKables 10' & 20' 6-channel snakes, Darktrain 2 & 4 channel KCY and mini xlr extensions:
Preamps:    Schoeps VMS 02iub, Naiant IPA, Sound Devices Mixpre6 I
Recorders: Sound Devices Mixpre6 I, Sony PCM m10

Home Playback: Mac Mini> Mytek Brooklyn+> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-4XX / Beyerdynamic DT880

Office Playback: iMac> Grace m903> AKG k701 / Hifiman HE-400

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #86 on: July 02, 2018, 03:35:21 PM »
Quote from: noahbickart link=topic=186648.msg2269858#msg2269858 date=1530410894[/quote

I’d be happy to walk you through it at some point, get in touch.

LOL, my friend, I need more than a walk through!  I need a "Posting For Dummies" printout...  I sat in with my friend Joe (not sure if you know him, but he goes by Joe Beacon).  He went through it with me step by step, and I wrote it all down.  I think I was able to get it to work once.  I think.  I may have a different version than him for something, but all it does is stress me out.  If you feel like dealing with it, and want to write out a step by step (taking nothing for granted), I'm a creature of habit and would follow the directions verbatim, and start posting lots of goodies.  I would only post to bt.etree, just so you know.  When I want something posted, I give it to Joe, but don't always like to ask because he has enough of his own stuff to deal with, so mine often doesn't get posted.

Well, I teach people how to understand the Talmud for a living, and I guarantee that bt.etree.org is less obtuse than the Babylonian Talmud....

OK, after reading this, I cracked up laughing... people at work turned to look at me!  You may have a bit of a challenge, but I'm game if you are.
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Dede2002

  • Trade Count: (0)
  • Taperssection All-Star
  • ****
  • Posts: 1217
  • Gender: Male
Re: What sample rate are you using?
« Reply #87 on: July 31, 2018, 08:44:01 PM »
Dsatz is one of the main reasons that Ts.com remains the best place on the internet.

Thanks for your continued presence here.

Same here. Dsatz posts are free humility lessons. It's one of the few opportunities in life where I feel exultant to find out how little I know.
Mics..........................SP-CMC-8, HLSC-1 and HLSO-MICRO
BB and Preamps........MM Micro bb / MM Custom Elite bb / Church 9100
                              
Recorders...................Tascam DR-100MKIII, Marantz PMD 620 MKII, Edirol R-09

Offline morst

  • I think I found an error on the internet; #UnionStrong
  • Trade Count: (2)
  • Needs to get out more...
  • *****
  • Posts: 5950
Re: What sample rate are you using?
« Reply #88 on: August 01, 2018, 02:02:59 PM »
LOL, my friend, I need more than a walk through!  I need a "Posting For Dummies" printout... 
Are you having trouble preparing a file set which is ready to share? Or having trouble with the uploading process on either archive.org or bitTorrent?


File set preparation is the same for either, but you would need a separate lesson for torrenting, since it's quite different than posting on the LMA.


It's not that any of the steps is that difficult, but it's a lot of ducks to get lined up in a row. It's great when you can get things to be "just so," and then have the upload work properly on the first try! The archive is much more forgiving than a torrent, in that you can change everything later. Torrents have to keep the fileset exactly as it is created in order to keep them running.

https://toad.social/@morst spoutible.com/morst post.news/@acffhmorst

Offline nak700s

  • Trade Count: (0)
  • Taperssection Member
  • ***
  • Posts: 634
  • Gender: Male
Re: What sample rate are you using?
« Reply #89 on: August 02, 2018, 04:19:03 PM »
LOL, my friend, I need more than a walk through!  I need a "Posting For Dummies" printout... 
Are you having trouble preparing a file set which is ready to share? Or having trouble with the uploading process on either archive.org or bitTorrent?


File set preparation is the same for either, but you would need a separate lesson for torrenting, since it's quite different than posting on the LMA.


It's not that any of the steps is that difficult, but it's a lot of ducks to get lined up in a row. It's great when you can get things to be "just so," and then have the upload work properly on the first try! The archive is much more forgiving than a torrent, in that you can change everything later. Torrents have to keep the fileset exactly as it is created in order to keep them running.

Where do I start, LOL????

If I had my step by step list here, I'd share it with you to help me troubleshoot, but I don't.  I use TLH to create the torrent, and everything "seems" to go just fine.  Once done, I attempt to upload to bt.etree, which I believe has worked once for me.  I am a creature of habit, and follow the same directions each time, so it doesn't make sense to me that it would work once and not again.  I only want to upload to bt.etree, so I keep it simple like that.  Basically, what I need is an idiot-proof step by step of how to create the torrent and then upload it to bt.etree.  Any help would be appreciated...
Normal: Nakamichi CM-700's >> SD 744T (or) Sony PCM-M10
Normal: Crown CM-700's >> SD 302 >> SD 744T
Fun times: 3 Crown CM-700's >> SD 302 >> SD744T + 2 Nakamichi CM-700's >> SD744T
Stealth: CA-14c >> CA 9200 >> Edirol R-09HR
Ultra stealth: AudioReality >> AudioReality battery box >> Edirol R-09HR
Simple & Sweet!

Offline Popmarter

  • Trade Count: (2)
  • Taperssection Member
  • ***
  • Posts: 738
  • Gender: Male
    • Live Music Archive
Re: What sample rate are you using?
« Reply #90 on: August 15, 2018, 02:19:08 PM »
24/48 Is enough
Recorders: Zoom F3; SD MixPre 3 II; Sony A10; Edirol R44; Sony M10; Sony WM-D6; iRiver IHP-120; Sharp MD-MT20; Sharp MD-MT190
Microphones: Schoeps MK41; Nevaton MC59/S (cards); Milab VM-44 Links (cards), AT853 7.4mod (cards); AT831 (cards); Nakamichi CM300 (all CP's); Soundman OKM II Rock Studios
Preamps: Beyerdynamic MV100; JK Laboratories DVC-X-17b; Naiant IPA; Nakamichi MX-100 modded for 9v battery use ; Baby Nbox

Offline Chris Damon

  • Trade Count: (0)
  • Taperssection Newbie
  • *
  • Posts: 3
Re: What sample rate are you using?
« Reply #91 on: August 22, 2018, 04:18:21 PM »
1bit/5.8 Mhz record and playback

Offline capnhook

  • All your llamas are belong to us....
  • Site Supporter
  • Trade Count: (20)
  • Needs to get out more...
  • *
  • Posts: 4843
  • All your llamas are belong to us....
Re: What sample rate are you using?
« Reply #92 on: August 22, 2018, 04:22:21 PM »
24/44.1

yep, but 24/48 if the video folks want some.

Switching to 24/48 for everything now, and using r8brain to resample.
Proud member of the reality-based community

BSCS-L->JB-mod [NAK CM-300 (CP-3) and/or (CP-1)]->LSD2->CA CAFS-Omni->Sony ECM-907**Apogee MiniMe Rev. C->CA Ugly II->**Edirol OCM R-44->Tascam DR-22WL->Sony TCD-D8


"Don't ever take an all or nothing attitude when it comes to making a difference
and being beautiful and making the world a beautiful place through your actions.
Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

"It'll never be over, 'till we learn."
 
"My dream is to get a bus and get the band and just go coast to coast. Just about everything else except music, is anti-musical.  That's it.  Music's the thing." - Jeb Puryear

Offline rigpimp

  • Trade Count: (14)
  • Needs to get out more...
  • *****
  • Posts: 3177
  • Gender: Male
  • Jarts don't kill people!
Re: What sample rate are you using?
« Reply #93 on: August 23, 2018, 10:38:38 AM »
Switching to 24/48 for everything now, and using r8brain to resample.

When I did re-sample I was also a r8brain user.  Excellent resampler!
Mics: Schoeps MK 5 MP, Schoeps MK 8 MP, Schoeps MK 41 MP, KCY 250/5 > PFA
Pre/A>D/P48: Sonosax SX/M2, Sonosax SX/M2-LS, E.A.A. PSP-2, Baby Nbox, Neumann BS48i-2 (for sale)
Recorders: Sound Devices Mixpre-6ii, Sony PCM-A10
Playback: Jolida 1501 Hybrid > McIntosh MX 130 > Von Schweikert VR-4 JR, or Little Dot MK III > Sennheiser HD700
http://archive.org/bookmarks/kskreider
https://www.concertarchives.org/kskreider
https://archive.org/details/thespps

Offline rocksuitcase

  • Trade Count: (4)
  • Needs to get out more...
  • *****
  • Posts: 8281
  • Gender: Male
    • RockSuitcase: stage photography
Re: What sample rate are you using?
« Reply #94 on: August 23, 2018, 11:35:47 AM »
1bit/5.8 Mhz record and playback
Oooooohh! You DSD people are soooooooooo singular!     >:D

Seriously, that is not a bad answer to the original topic. Do you have the Korg MR1 (2?) which some other guys use? keo, a non TS'er local taper in my area and raoulduke have them.
I've always been interested in the technology but never been convinced enough to go for it.

Keep posting Chris!

music IS love

When you get confused, listen to the music play!

Mics:         AKG460|CK61|CK1|CK3|CK8|Beyer M 201E|DPA 4060 SK
Recorders:Marantz PMD661 OADE Concert mod; Tascam DR680 MKI x2; Sony PCM-M10

Offline Chuck

  • Trade Count: (42)
  • Needs to get out more...
  • *****
  • Posts: 10810
  • Gender: Male
  • time between the notes...
    • My recordings on the LMA
Re: What sample rate are you using?
« Reply #95 on: August 24, 2018, 01:57:18 PM »
I stumbled onto this thread and was pleasantly surprised to see that DSatz has arrived at the same conclusion I have on appropriate bit rate and sample size for recording music. Since I bought my MixPre-6 I've started recorded everything at 24 bit/44.1 kHz. I'm a recent and somewhat reluctant convert to 24 bit, because the math always says 16 bits are enough. I don't think recording at a higher sample frequency is worthwhile.

What I'm still trying to figure out is how to properly resample from 24 bit to 16 bit. I've read a lot about this and still haven't figured out how best to do that. For now I'm using UV22HR dither simply because it's built into CuBase which I use for editing. I've read all the arguments, some say don't dither, because it adds noise etc... But, just chopping those bits off doesn't sit well with me either.

I always listen at 16 bit/44.1 kHz.

« Last Edit: August 24, 2018, 04:48:01 PM by Chuck »
Reality is merely an illusion, albeit a very persistent one.

Microphones: AKG C 480 B comb-ULS/ CK 61/ CK 63, Sennheiser MKE 2 elements,  Audix M1290-o, Micro capsule active cables w/ Naiant PFA's, Naiant MSH-1O, Naiant AKG Active cables, Church CA-11 (cardioid), (1) Nady SCM-1000 (mod)
Pre-amps: Naiant littlebox, Naiant littlekit v2.0, BM2p+ Edirol UA-5, Church STC-9000
Recorders: Sound Devices MixPre-6, iRiver iHP-120 (Rockboxed & RTC mod)

Recordings on the LMA: http://www.archive.org/bookmarks/ChuckM
Recording website & blog: http://www.timebetweenthenotes.com

Offline Gutbucket

  • record > listen > revise technique
  • Trade Count: (15)
  • Needs to get out more...
  • *****
  • Posts: 15700
  • Gender: Male
  • "Better to love music than respect it" ~Stravinsky
Re: What sample rate are you using?
« Reply #96 on: August 24, 2018, 04:39:52 PM »
What I'm still trying to figure out is how to properly resample from 24 bit to 16 bit. I've read a lot about this and still haven't figured out how best to do that. For now I'm using UV22HR dither simply because it's built into CuBase which I use for editing. I've read all the arguments, some say don't dither, because it adds noise etc... But, just chopping those bits off doesn't sit well with me either.

You are not actually resampling when you make that conversion. Resampling is changing the sampling rate- converting from 48kHz to 44.1kHz for example.  Resampling is more computationally complex than changing bit rate (dithering and truncating), and IMO there is a stronger argument for using a high-quality resampling routine to do so (or better, recording at the target rate to begin with) than there is for the use of noise-shaped dithers like UV22HR or whatever when reducing bit length, especially given the noise floor of the material most live music tapers are recording.


Here's my take on changing bit length-
tldr- Any time you change from a higher bit rate to a lower one, apply dither as part of the truncation process.  Both are done as essentially a single step in your editor. 

Dither is essentially very quiet randomized (uncorrelated) noise.  Don't let the term "noise" scare you.  This is very, very quiet noise which you can only hear as hiss if you were to crank the volume to ungodly levels which would otherwise destroy your system and eardrums when normal level program material plays.  It only affects the least-significant bit.  By applying it you trade one type of noise for another.  You eliminate one type of very unmusical, nasty sounding, extremely quiet digital artifact noise for a far less annoying, but also extremely quiet analog hiss.  In addition, dither allows super, super quiet fading sounds to still be heard as they sink beneath that analog like hiss noise-floor, rather than suddenly dropping off into digital silence (and creating that nasty digital artifact noise as they do so).  But again, you'd only actually hear this if you were to crank up the volume to ungodly levels.

Whenever you reduce the bit-length of the file you raise its lowest achievable (in our case theoretical) noise floor.  But the lowest achievable noise floor of the file is almost never the actual noise-floor of any of our live recordings.  Instead they are always dominated by other forms of noise at higher levels.  If your recording gain is set to anything reasonable, the self-noise of the microphones will always be higher than the mathematical lower limit of the file.  Self-noise of microphones is basically random hiss, which essentially serves as dither in its own right.  So even if you chose not to dither when shortening the bit length, your file is still probably dithered with plenty of mic-noise at the bottom.  So why dither?  Because it can't hurt and does no harm.  It's good practice and takes no more effort than truncating without dither.

And for the vast majority of live-music, the noise floor of the recording is dominated not by mic self-noise but by the acoustic noise floor of the environment in which the recording was made.  This noise-floor is likely to be 30dB or more above the lower limit of even a 16 bit file.  Dither noise is way, way quieter and will never be heard in these live recordings even if you crank it up.  The HVAC noise of the room and people breathing and fidgeting in even super quiet classical venue recordings will dominate.  Fugetaboutit with anything amplified through a PA.

I use standard "triangular" dither, which should be an option in any editor.  Noise shaped dithers like UV22HR are fine and won't hurt for what we are doing, but also probably make no difference, after all it's going to be buried deep beneath other noise.  The idea behind noise shaped dithers is that the dither noise is not linear with frequency like standard dither noise but rather EQ'd so that there is more noise where the ear is less sensitive (low and very high frequencies) and less where the ear is most sensitive.  Okay that's cool, and may arguable be useful with super-duper quiet recordings made under controlled conditions, but again, if its buried beneath higher level noise it won't matter. 

Actually there is a technical argument against using noise-shaped dither for sources which may be mixed together again.  That's because the specially shaped non-linear noise adds together and is then no longer the specially crafted shape anymore, it becomes over-emphasized.  Best practice would be to do all the mixing at higher bit lengths and dither/truncate as a last step. Next best is probably to use standard dither for each source if saving them separately at a lower bit length prior to mixing, then maybe choose noise-shaped dither if one wanted to as the final step after they've been mixed if reducing the bit-rate still further.  But this is all gilding the lily and seems to me a solution without a problem for post production of live music recordings.  Noise shaped dithers like UV22HR, SBM, etc, arguably have more usefulness in a 24bit capable ADC's which is doing the bit reduction prior to recording a 16 bit file.  The argument there is basically the same as that for recording in 24 bits instead of 16, providing a touch more leeway in setting recording levels safely between noise at the bottom and overs at the top, but not as much as recording at 24 bits to begin with.
musical volition > vibrations > voltages > numeric values > voltages > vibrations> virtual teleportation time-machine experience
Better recording made easy - >>Improved PAS table<< | Made excellent- >>click here to download the Oddball Microphone Technique illustrated PDF booklet<< (note: This is a 1st draft, now several years old and in need of revision!  Stay tuned)

Offline Chuck

  • Trade Count: (42)
  • Needs to get out more...
  • *****
  • Posts: 10810
  • Gender: Male
  • time between the notes...
    • My recordings on the LMA
Re: What sample rate are you using?
« Reply #97 on: August 24, 2018, 06:16:36 PM »
Thanks Gutbucket. You've conformed my thoughts.

Great informational thread here.
Reality is merely an illusion, albeit a very persistent one.

Microphones: AKG C 480 B comb-ULS/ CK 61/ CK 63, Sennheiser MKE 2 elements,  Audix M1290-o, Micro capsule active cables w/ Naiant PFA's, Naiant MSH-1O, Naiant AKG Active cables, Church CA-11 (cardioid), (1) Nady SCM-1000 (mod)
Pre-amps: Naiant littlebox, Naiant littlekit v2.0, BM2p+ Edirol UA-5, Church STC-9000
Recorders: Sound Devices MixPre-6, iRiver iHP-120 (Rockboxed & RTC mod)

Recordings on the LMA: http://www.archive.org/bookmarks/ChuckM
Recording website & blog: http://www.timebetweenthenotes.com

 

RSS | Mobile
Page created in 0.587 seconds with 126 queries.
© 2002-2024 Taperssection.com
Powered by SMF