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Author Topic: What sample rate are you using?  (Read 4129 times)

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Offline checht

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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
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Offline Fatah Ruark (aka MIKE B)

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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.
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Offline rocksuitcase

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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.
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Offline heathen

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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.
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Offline rigpimp

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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.
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Offline firemt66

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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

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Re: What sample rate are you using?
« Reply #6 on: June 03, 2018, 08:42:50 AM »
24/48
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Offline kindms

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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
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Offline wforwumbo

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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.
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Offline Gordon

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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.
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Offline ilduclo

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Re: What sample rate are you using?
« Reply #10 on: June 03, 2018, 10:40:21 AM »
24/44.1

Offline Sloan Simpson

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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.
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Offline ilduclo

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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

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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.

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Offline billydee

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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.

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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(?)
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Offline Scooter123

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Re: What sample rate are you using?
« Reply #16 on: June 03, 2018, 12:13:30 PM »
24 48

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Offline jagraham

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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.
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Offline capnhook

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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.




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"Don't ever take an all or nothing attitude when it comes to making a difference
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Every little bit is registered.  Every little bit.  So be as beautiful as you can as often as you can"

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Offline checht

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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'.
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Offline capnhook

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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.






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"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."
 
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Offline checht

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Re: What sample rate are you using?
« Reply #21 on: June 03, 2018, 05:12:03 PM »
^ Thanks!
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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

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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.
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Offline nulldogmas

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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

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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.
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Offline ycoop

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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?
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Offline aaronji

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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

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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

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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
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"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

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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.
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Re: What sample rate are you using?
« Reply #31 on: June 05, 2018, 01:02:01 AM »
24/44.1
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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)_/¯
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Offline Gutbucket

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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.
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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.
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Offline IMPigpen

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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.
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Offline DSatz

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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 »
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Offline morst

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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!)
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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?

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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.

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Offline DSatz

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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

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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.
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Offline checht

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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?

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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.
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Offline DSatz

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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

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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.
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Offline aaronji

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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...

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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
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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
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Offline wforwumbo

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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.
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Offline goodcooker

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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:
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Offline Gutbucket

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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 »
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Offline checht

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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.
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Offline wforwumbo

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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.
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Offline checht

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Re: What sample rate are you using?
« Reply #54 on: June 14, 2018, 12:52:41 AM »
^ That clarifies a lot.

Thanks!
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Offline Gutbucket

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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.
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Offline DSatz

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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 »
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Offline wforwumbo

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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.
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Offline nak700s

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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!
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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

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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...

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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.
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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.

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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.
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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?

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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.

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Offline beenjammin

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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?

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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.
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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?
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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

Offline beenjammin

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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

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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.
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Offline beenjammin

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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

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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?
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Offline Gutbucket

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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.
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Offline nak700s

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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
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Offline Gutbucket

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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.

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Offline Gutbucket

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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.
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Offline nak700s

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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
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Offline noahbickart

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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), mk4v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, Darktrain 2 and 4 channel KCY extensions:
Preamps:    Naiant Littlebox, Naiant IPA, Naiant PFA, Sound Devices Mixpre6
Recorders: Sound Devices Mixpre6, Sony PCM m10
Home Playback: Mytek DSD 192> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-400
Office Playback: Grace m903> AKG k701

Offline Gutbucket

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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
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Offline Gutbucket

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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 »
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Offline nak700s

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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

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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), mk4v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, Darktrain 2 and 4 channel KCY extensions:
Preamps:    Naiant Littlebox, Naiant IPA, Naiant PFA, Sound Devices Mixpre6
Recorders: Sound Devices Mixpre6, Sony PCM m10
Home Playback: Mytek DSD 192> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-400
Office Playback: Grace m903> AKG k701

Offline nak700s

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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

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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), mk4v (x2), mk22 (x2), mk3 (x2), mk21 & mk8
Cables: 2x nbob KCY, 1 pair nbob actives, Darktrain 2 and 4 channel KCY extensions:
Preamps:    Naiant Littlebox, Naiant IPA, Naiant PFA, Sound Devices Mixpre6
Recorders: Sound Devices Mixpre6, Sony PCM m10
Home Playback: Mytek DSD 192> McIntosh MC162> Eminent Tech LFT-16; Musical Fidelity xCan v2> Hifiman HE-400
Office Playback: Grace m903> AKG k701

Offline nak700s

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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

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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.
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Offline morst

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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.

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Offline nak700s

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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...
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Offline Popmarter

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Re: What sample rate are you using?
« Reply #90 on: August 15, 2018, 02:19:08 PM »
24/48 Is enough
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