So I have about 22 tracks transfered from a Tascam 24 track 1" tape source into 24-bit/44.1KHz and I am doing a mix.
I was thinking, having never ventured above 44.1KHz if it would be an advantage to process and mixdown stems in a higher sample rate. I will only be using SFP effects, and I will do like the toms, kick and snare etc. in separate go's as I only have 12DSPs.
Then in the end I will mix the stems down to a stereo mix.
It would be cool to try, even though it will probably end up on CD, or possibly vinyl.
Or perhaps only 48KHz?
What is your opinion?
Benefits of prcessing 44.1KHz data in 88.2KHz/96KHz?
Re: Benefits of prcessing 44.1KHz data in 88.2KHz/96KHz?
imho not a real advantage
going to other sample frequences but 88.2KHz will compromise the quality of audio data due to conversion approximation.
Going to 88.2 supposedly (depending on the way you do it) requires only sample duplication which doesn't involve sample amplitude re-shaping (noise adding). In such a case you'll have no loss. Further processing done at 88KHz will work within an extended frequency response from 20 to 44100 (instead of 20-22050): if your gear (player, audio amplifier, tweeters... ears) supports such a bandwidth, as result you'll hear effetcs processing also in the range between 22050Hz and 44100Hz which was impossible to hear with the 44.1KHz sampling frequency.
If you plan to consequently downsample to 44.1KHz you'll probably (in the best case) obtain the same result as doing nothing.
All this not taking into account possible software/encoding bugs.
I also suppose that working with double frequency will half the dsp processing power.
cheers Fede
going to other sample frequences but 88.2KHz will compromise the quality of audio data due to conversion approximation.
Going to 88.2 supposedly (depending on the way you do it) requires only sample duplication which doesn't involve sample amplitude re-shaping (noise adding). In such a case you'll have no loss. Further processing done at 88KHz will work within an extended frequency response from 20 to 44100 (instead of 20-22050): if your gear (player, audio amplifier, tweeters... ears) supports such a bandwidth, as result you'll hear effetcs processing also in the range between 22050Hz and 44100Hz which was impossible to hear with the 44.1KHz sampling frequency.
If you plan to consequently downsample to 44.1KHz you'll probably (in the best case) obtain the same result as doing nothing.
All this not taking into account possible software/encoding bugs.
I also suppose that working with double frequency will half the dsp processing power.
cheers Fede
like said above, but also i think 24/44100 is already very nice high quality.
scope plugs do sound great on 96 though, specially the vectron blows me away on 96... but this is going a bit offtopic
scope plugs do sound great on 96 though, specially the vectron blows me away on 96... but this is going a bit offtopic
Scope, Android, Web, PC Plugins and Sounds:
http://www.oceanswift.net
Music
https://faxinadu.bandcamp.com/
http://www.oceanswift.net
Music
https://faxinadu.bandcamp.com/
I think the "88.2K is better for 44.1K"-argument is mere myth actually as sample-conversion is not just about dropping every other sample. But 88.2K should surely be enough.
The reason I want to try this is because most SFP effects other than the high-end synths have no over-sampling, so I would like to try this and then apply a good SRC (like r8brain) to the final mix for a steep, low-aliasing filter.
The reason I want to try this is because most SFP effects other than the high-end synths have no over-sampling, so I would like to try this and then apply a good SRC (like r8brain) to the final mix for a steep, low-aliasing filter.
For recording, it doesn't matter much. Upsampling drumsamples is probably a waste. But for synthesis (pitching, oscillators, filters), EQ's and stuff, you can clearly hear the difference between 44.1 and 96kHz. If it doesn't cause trouble for the rest of the system, and you're after that quality, why not go for it? That's probably why your "vectron blows you away on 96"...faxinadu wrote:[...] but also i think 24/44100 is already very nice high quality.
scope plugs do sound great on 96 though, specially the vectron blows me away on 96... but this is going a bit offtopic
more has been done with less
https://soundcloud.com/at0m-studio
https://soundcloud.com/at0m-studio
think about what makes the differnce between 44.1 and 96 ...
aliasing artifacts are moved beyond the audible range of the spectrum
which makes a cleaner, more transparent and somewhat smoother sound
it's in no way at all related to a 'more datapoints - more precision' assumption
as aliasing is a 'symmetric' (so to say) relation between the source frequency and the sampling frequency, it's kind of predictable - and as such a proper 'arrangement' of the track's spectral content should be able to cover most of it's unwanted presence.
An experienced engineer probably does exactly that by intuition.
the Vectron is a special case as it's using 8bit wavetables - the original Waldorf tables included in SFP are like that, aren't they ?
and that sound is intended to alias by nature ....
due to the nature of that nature a high frequency processing is rather spectacular, but is it really 'better' (?) ...which is of course a matter of taste anyway.
some of the best SFP synth patches feature theWaldorf Oscillators, and guess what makes the filters really shine... what should they mangle if there was only a dead precise note ?
I love aliasing - I've recently patched a Masterverb and an external unit parallel (to a/b between them) in the FX path of my guitar.
The MV is lightyears ahead in precision and transparency (I find it pretty useful for guitar processing), but it just cannot create the dense melted atmosphere of the 12 bit 31k Boss RRV-10 (a breed between trash and gorgeous) .
Not all of it's sounds are usable all the time, but when you dig a matching one, it's mindblowing
my bottomline would be to pay attention to the individual sounds and anticipate critical overlays that may be particulary sensible to the nasty parts of aliasing, but otherwise use the full spectrum. Who said it would be easy anyway ?
cheers, Tom
aliasing artifacts are moved beyond the audible range of the spectrum
which makes a cleaner, more transparent and somewhat smoother sound
it's in no way at all related to a 'more datapoints - more precision' assumption
as aliasing is a 'symmetric' (so to say) relation between the source frequency and the sampling frequency, it's kind of predictable - and as such a proper 'arrangement' of the track's spectral content should be able to cover most of it's unwanted presence.
An experienced engineer probably does exactly that by intuition.
the Vectron is a special case as it's using 8bit wavetables - the original Waldorf tables included in SFP are like that, aren't they ?
and that sound is intended to alias by nature ....
due to the nature of that nature a high frequency processing is rather spectacular, but is it really 'better' (?) ...which is of course a matter of taste anyway.
some of the best SFP synth patches feature theWaldorf Oscillators, and guess what makes the filters really shine... what should they mangle if there was only a dead precise note ?
I love aliasing - I've recently patched a Masterverb and an external unit parallel (to a/b between them) in the FX path of my guitar.
The MV is lightyears ahead in precision and transparency (I find it pretty useful for guitar processing), but it just cannot create the dense melted atmosphere of the 12 bit 31k Boss RRV-10 (a breed between trash and gorgeous) .
Not all of it's sounds are usable all the time, but when you dig a matching one, it's mindblowing
my bottomline would be to pay attention to the individual sounds and anticipate critical overlays that may be particulary sensible to the nasty parts of aliasing, but otherwise use the full spectrum. Who said it would be easy anyway ?
cheers, Tom
This has been covered before:
Another reason that 44.1 is slightly frowned upon is because in the 'professional' world of the 80's and early 90's when digital technology for recording started to proliferate, most people still mixed on outboard gear.
This has been covered before:
44.1khz is a really bare comprimise when it comes to moving the effects of the filter topology required for the DAC's beyond the limits of human hearing. 22050 either requires a shallower filter set lower (16-18khz) or a really steep filter set at 20khz (which increases ringing but improves transients quite a bit as well). Many units use analog filters on the output side & digital filters on the input side (FIR filters) to achieve the best performance for each given task. The rest of the ADC or DAC's component design and the quality of your clock also affect the sound of course, but this is true regardless of samplerate.
So the reason for the refresher (which I hope is at least mostly correct at 4:20am ) is because when people in the beginning of the digital recording era used converters it was often multiple generations. 48khz put the filter at a higher frequency and allowed even more transparent shallower (less phase distorted) recordings, and the theory goes one or two passes at that samplerate then mixed in analog (or :shudder: transcoded) to 44.1 for the final master allowed for less combined effects of the filters.
Now people claim that recording at 96khz (or 192khz and so on) reduces this effect even more in addition to reducing the above mentioned aliasing effects that occur with the use of digital processes (filter/eq/etc) when mixing in the digital realm, or calculating synthesis etc in the digital realm.
So, using a higher samplerate might make the recorded material a bit more transparent and it might improve the quality of the eq & synthesis in your mixing chain, but if you're not high end botique recording channels (Chandler/Neve/SSL/Crane) alongside your shakti stones & oversized wooden knobs, is the 'low' samplerate really the worst thing in your signal chain?
It's also worth repeating Bob Katz & Massenburg etc who have all stated that a well designed ADC or DAC working at 44.1 will sound far better than an awful design working at 96khz.
Another reason that 44.1 is slightly frowned upon is because in the 'professional' world of the 80's and early 90's when digital technology for recording started to proliferate, most people still mixed on outboard gear.
This has been covered before:
44.1khz is a really bare comprimise when it comes to moving the effects of the filter topology required for the DAC's beyond the limits of human hearing. 22050 either requires a shallower filter set lower (16-18khz) or a really steep filter set at 20khz (which increases ringing but improves transients quite a bit as well). Many units use analog filters on the output side & digital filters on the input side (FIR filters) to achieve the best performance for each given task. The rest of the ADC or DAC's component design and the quality of your clock also affect the sound of course, but this is true regardless of samplerate.
So the reason for the refresher (which I hope is at least mostly correct at 4:20am
) is because when people in the beginning of the digital recording era used converters it was often multiple generations. 48khz put the filter at a higher frequency and allowed even more transparent shallower (less phase distorted) recordings, and the theory goes one or two passes at that samplerate then mixed in analog (or :shudder: transcoded) to 44.1 for the final master allowed for less combined effects of the filters.Now people claim that recording at 96khz (or 192khz and so on) reduces this effect even more in addition to reducing the above mentioned aliasing effects that occur with the use of digital processes (filter/eq/etc) when mixing in the digital realm, or calculating synthesis etc in the digital realm.
So, using a higher samplerate might make the recorded material a bit more transparent and it might improve the quality of the eq & synthesis in your mixing chain, but if you're not high end botique recording channels (Chandler/Neve/SSL/Crane) alongside your shakti stones & oversized wooden knobs, is the 'low' samplerate really the worst thing in your signal chain?
It's also worth repeating Bob Katz & Massenburg etc who have all stated that a well designed ADC or DAC working at 44.1 will sound far better than an awful design working at 96khz.
definetely not, as I could experience just a couple of hours before I wrote my post above...valis wrote:..., is the 'low' samplerate really the worst thing in your signal chain? ...
Since I've learned that the whole signal chain (from instrument to cabinets) matters, I took my bass to the gear shop to make some comparisons.
holy sh*t - I didn't expect such a difference
I favoured a certain combo (fr. prev test w. one of the shop's instruments), but now with my own it completely s*cked, not somewhat, but BIG - and it was a 1k Euro item
I also knew from previous test that an Ashdown 300W top was pretty versatile in sound adjustment - with a H&K 4x10" cabinet I got 'my sound' - yeah, that was it
the funny thing (as I'm not a good player at all), is that this sound and feeling immediately fired some inspiration and made the fingers run and plug almost by themselves - a sure sign that it's good as it is
now checking the (matching) H&K top (also 300W) with the same cabinet.
oops, what's up ??? where's the smooth round bottom plus a bit of sparkling brilliance ?It didn't exactly s*ck (as the first combo), but I found it a bit limited, big attack and 'flat' forward, I would say - but what a tremendous difference in sound character by just changing the amp. (all gear was in a similiar price range btw and no extensive eqs applied)
what was the question about what rate... ?
cheers, Tom