I have read good things about Voxengo R8brain Pro. It´s a software sample rate converter. There is a Demo version limited to 1 min of lengh. If is true what is told about it, it must be a very good converter.
My own experience is that 96k audio recording seems a bit better on high frequencies. That´s what I have found with my Rosetta AD 96k. Probably, new Apogee AD16x converters at 192k, may be nearer the analog sound, but haven´t heard it.
Regarding to Synths, sampling, and VSTi, realize many sofware is made to run at 44.1, and sounds worse at 96khz. There are some VSTi where occurs. Also, most of sampling libraries are at 44.1.
This thread have making me to remember one idea. If could be recording on both sample rates at time, on one machine.
I mean, run two apps, syncronized anyway (system link, midi clock), running on two different soundcards, one at 44.1, the other at 96k. The first for sampling, synths, instruments, the second for audio recording.
Do you think is posible? useful?
The idea would be mix separately, and get two mixdowns. And final mix this two ones at the same sample rate.
the samplerate (test) thread
astroman, you were correct - a.wav is an original 44.1kHz, while b.wav is resampled from 96kHz. For whom it may be unclear - check out the overtones and rattling in the higher frequency (aliasing) that occurs at 44.1kHz oscillator in a.wav. The b.wav may have some artefacts from the basic resampling function in SX, but still, it's overtones are purer, rounder as someone mentions. The artefacts from the 96kHz synth happened mainly around 48 and 24kHz, both outside the audible range.
More recently, some artists (Vitalic anyone?) use aliasing very musically, but still, it's a side-effect you don't want when synthesising virtual strings or Moog emulations.
I think it was garyb that posted an excellent link that explained aliasing, by an example we all know well: the wheels of a car sometimes appear to be spinning in reverse on TV. That's what aliasing is.
For recording, samplerate doesn't matter as long as you're not studying bats
re: VSTi
With VSTi it's a different thing, since many programmers make an internal oversamling to make the synths sounds better. Also, programs like Fruityloops and Ableton have rendering options that will do much the same, but is way slower than realtime playback/synthesis. Who cares if it takes the program 30 minutes to render a 3 minute song if it sounds way better than just recording the realtime output.
More recently, some artists (Vitalic anyone?) use aliasing very musically, but still, it's a side-effect you don't want when synthesising virtual strings or Moog emulations.
I think it was garyb that posted an excellent link that explained aliasing, by an example we all know well: the wheels of a car sometimes appear to be spinning in reverse on TV. That's what aliasing is.
For recording, samplerate doesn't matter as long as you're not studying bats
re: VSTi
With VSTi it's a different thing, since many programmers make an internal oversamling to make the synths sounds better. Also, programs like Fruityloops and Ableton have rendering options that will do much the same, but is way slower than realtime playback/synthesis. Who cares if it takes the program 30 minutes to render a 3 minute song if it sounds way better than just recording the realtime output.
more has been done with less
https://soundcloud.com/at0m-studio
https://soundcloud.com/at0m-studio
ok guys, this discussion leads me to the next question marks:
1. If i were able to do my whole processing path from recording to mastering with 96khz, does the creamware mixers, effects, etc. work internally with 96khz or are they limited?
2. Are VST-Plugs, i.e. waves, fully capable of working with 96khz?
greez
roman
1. If i were able to do my whole processing path from recording to mastering with 96khz, does the creamware mixers, effects, etc. work internally with 96khz or are they limited?
2. Are VST-Plugs, i.e. waves, fully capable of working with 96khz?
greez
roman
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lagoausente
- Posts: 552
- Joined: Wed Feb 09, 2005 4:00 pm
- Location: Spain
I can't tell about the tech stuff as good as other people here. But my ears are pretty consistent in choosing MasterwerbPro in 96KHz over 44.1KHz. Another thing is, that some people do "sample rate conversion" by going D/A out of a 96KHz system and A/D into a 44.1KHz system. If you use outboard analog compression anyway, then the "96 -> 44.1 makes trouble" talk is irrelevant anyway.
ok, i tested a bit yesterday how far i could go with 96khz...
my config again: 2x PowerPulsar II (15 DSP's each), connected with 5 STDM cables
1. I started with an empty project
2. I set the project to 96khz (scope as master)
3. I loaded a STM2448 mixer
4. I loaded the ASIO2-24bit module
5. I set the module to 24 channels (in and out)
6. I loaded the zlink source
7. I connected the 16 zlink to ASIO-Dest
8. Then I began to connect the ASIO-Source to the STM2448
9. While connecting the 21st channel i got the "big modules does no fit"
10. I removed the 21 connection to the mixer
11. DSP load was about 30%
12. I loaded a Masterverb with no problems
13. I unloaded the Masterverb
14. I loaded the Masterverb Pro and got the "big modules does not fit" again...
15. I tried to set scope to slave, to a dbx376 channel strip, got the "big modules" again.
16. stopped testing and went to the fridge for a beer
cheerz
roman
my config again: 2x PowerPulsar II (15 DSP's each), connected with 5 STDM cables
1. I started with an empty project
2. I set the project to 96khz (scope as master)
3. I loaded a STM2448 mixer
4. I loaded the ASIO2-24bit module
5. I set the module to 24 channels (in and out)
6. I loaded the zlink source
7. I connected the 16 zlink to ASIO-Dest
8. Then I began to connect the ASIO-Source to the STM2448
9. While connecting the 21st channel i got the "big modules does no fit"
10. I removed the 21 connection to the mixer
11. DSP load was about 30%
12. I loaded a Masterverb with no problems
13. I unloaded the Masterverb
14. I loaded the Masterverb Pro and got the "big modules does not fit" again...
15. I tried to set scope to slave, to a dbx376 channel strip, got the "big modules" again.
16. stopped testing and went to the fridge for a beer
cheerz
roman
you were absolutely correct in your bottomline
some things just make sense - others, not so much...
back to the topic regarding the difference between 6 and 14/15 DSP cards:
imho the source of the problem is not in the cards itself (or how they are built/adressed).
A smaller card is just less likely to receive enough 'problematic' modules in a typical project.
With only 'bummer' modules it will probably fail as soon as the other one (imho).
The effect of the higher samplerate (as demonstated with At0m's A/B files) will be significantly less obvious on (acoustic material) audio tracks.
In some cases it may not even be apprecited at all, as a 'too sweet' signal often benefits from some grain - 12 bit drumboxes, old Lexicons, exciters... anyone ?
I know at least one quote from a hardcore audiophile who tweaked his system to a precision close to the limits of measurability - and his conclusion was '... it s*cked completely, a sterile and lifeless tone...'
A great voice and an extraordinairy performance on an acoustic instrument deserve the 'best' conservation, but what's usually applied in postprocessing wipes out the small gain in quality between the 2 samplerates big times (imho).
A 96k Beyouncee will never 'sound' human, regardless how good looking she is...
I had the strange pleasure to listen to the voice of Marshall Matters for a short moment when the 'Eminem sound engine' failed during a live performance...
And what's pressed on Katie Melua CDs is a shame for the producer, a certain Mike Batt.
[edit]
that may be misunderstandable, as those CDs have a perfect soundquality, but the girl's beautiful voice is entirely drowned in reverb - shudder. It could be 384 khz and would still s*ck, on a high level, tho...
I happened to watch her sing live at an Echo Award (or something like that) and was really impressed.
[/edit]
Imho you get distracted by technology (#figures - like the American bigger-is-better quote above) if focussing too much on the samplerate. There are other domains in which the human ear (even the untrained) is much more sensible
my 2 cents, Tom
<font size=-1>[ This Message was edited by: astroman on 2006-08-23 10:05 ]</font>
some things just make sense - others, not so much...
back to the topic regarding the difference between 6 and 14/15 DSP cards:
imho the source of the problem is not in the cards itself (or how they are built/adressed).
A smaller card is just less likely to receive enough 'problematic' modules in a typical project.
With only 'bummer' modules it will probably fail as soon as the other one (imho).
The effect of the higher samplerate (as demonstated with At0m's A/B files) will be significantly less obvious on (acoustic material) audio tracks.
In some cases it may not even be apprecited at all, as a 'too sweet' signal often benefits from some grain - 12 bit drumboxes, old Lexicons, exciters... anyone ?
I know at least one quote from a hardcore audiophile who tweaked his system to a precision close to the limits of measurability - and his conclusion was '... it s*cked completely, a sterile and lifeless tone...'
A great voice and an extraordinairy performance on an acoustic instrument deserve the 'best' conservation, but what's usually applied in postprocessing wipes out the small gain in quality between the 2 samplerates big times (imho).
A 96k Beyouncee will never 'sound' human, regardless how good looking she is...
I had the strange pleasure to listen to the voice of Marshall Matters for a short moment when the 'Eminem sound engine' failed during a live performance...
And what's pressed on Katie Melua CDs is a shame for the producer, a certain Mike Batt.
[edit]
that may be misunderstandable, as those CDs have a perfect soundquality, but the girl's beautiful voice is entirely drowned in reverb - shudder. It could be 384 khz and would still s*ck, on a high level, tho...
I happened to watch her sing live at an Echo Award (or something like that) and was really impressed.
[/edit]
Imho you get distracted by technology (#figures - like the American bigger-is-better quote above) if focussing too much on the samplerate. There are other domains in which the human ear (even the untrained) is much more sensible
my 2 cents, Tom
<font size=-1>[ This Message was edited by: astroman on 2006-08-23 10:05 ]</font>
For recording & playback higher samplerates have the net effect of pushing the artifacts of conversion (filter comprimises mostly) beyond the limits of hearing but paying attention to clockspeed ALONE will NOT necessarily yield benefits. A well designed 44.1 converter will outperform a cheap soundcard of any samplerate every time.
Pay attention to at0m's words, the biggest gains are in algorithms such as oscillator & filter components (this INCLUDES eq's) and these are often oversampled internally anyway.
Pay attention to at0m's words, the biggest gains are in algorithms such as oscillator & filter components (this INCLUDES eq's) and these are often oversampled internally anyway.
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hubird
katano, Astroman and Valis (and Atom), great stuff for my simple mind, thanks!
Specially the difference between audio and synths related to samplerate stuff.
I was amazed by Atom's two wav examples, quite a difference.
(And I actually didn't like the resulting sound of both of them).
The above explanation(s) reassure me again, there's nothing I can/should/have to do today about the subject except giving my best attention to the sound of my tracks
(not even ironic).
Thanks guys
Specially the difference between audio and synths related to samplerate stuff.
I was amazed by Atom's two wav examples, quite a difference.
(And I actually didn't like the resulting sound of both of them).
The above explanation(s) reassure me again, there's nothing I can/should/have to do today about the subject except giving my best attention to the sound of my tracks
(not even ironic).
Thanks guys
You can do the test easily yourself: take a synth like Inferno, and select the "Reset All" preset. It has a saw oscillator (lots of overtones = high freq's), no disto, no subosc or ringmod, no pulse width or filter. Change samplerates and listen.On 2006-08-23 16:13, hifiboom wrote:
Still I would like to hear the original 96Khz sample to see if the difference may just is audible due to the converting from 96kHz to 48Khz/44,1khz...
more has been done with less
https://soundcloud.com/at0m-studio
https://soundcloud.com/at0m-studio