cheater
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Using S8 with sample rates higher than the hardware supports? Oversampling the project in Sonar 8?
Hi guys,
as many people I am for the time being stuck with a card that only supports 44.1 or 48kHz audio rates. However, at those settings, many my VSTi's really sound very bad because of the aliasing. Is there a way to set S8 to oversample the whole project, e.g. by 2x or 4x so that the whole audio path works at 192 kHz and then, at output, gets downsampled to 48k?
Alternatively, does anyone of you cool guys know of a Windows application/driver (that's win xp I'm using) that would 'fake' support, so that it shows Sonar 192k compatibility, and therefore lets me select that option with Sonar? I guess such a driver would then, again, downsample the audio and output it to my usual ASIO or WDM driver.
This also gets me a bit worried about project portability. If someone makes a song at 192kHz, and then goes to a studio that only has 48k outputs, the project could end up sounding much much worse - that's not very good! Or if, let's say, I'm collaborating with a friend and he has 192k while I only have 44.1 and I end up turning off many of his sounds in the mix because they start sounding very, very bad and even dysharmonic..
Please help guys! I know you are able to!
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pwal
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 06:57:10
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CJaysMusic
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 07:23:49
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44.1Khz or 48Khz should be good enough. 192Khz is overkill and you will not hear the difference unless your part dog. Record in 48 or 44 and enable the 64bit floating point engine.
Maybe your sound cards D/A converter sucks, cause 48Khz is more than enough. In my opinion, 96Khz is overkill and a waste of disk space and resources
Cj
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altima_boy_2001
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 07:46:37
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This also gets me a bit worried about project portability. People really worried about portability (especially with soft synths) will bounce tracks down to wave files rather than trying to get multiple DAW installations to work identically with each other. As for the other point, can the best sample rate conversion algorithms work in real time on a CPU (ie. without hardware support)? I don't know so I'm asking...That would be a requirement, but I suppose it's possible if they have oversampler plugins available...
You can use me as your eSoundz referral (altima_boy_2001).
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cheater
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 09:01:11
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ORIGINAL: CJaysMusic
44.1Khz or 48Khz should be good enough. 192Khz is overkill and you will not hear the difference unless your part dog. Record in 48 or 44 and enable the 64bit floating point engine.
Maybe your sound cards D/A converter sucks, cause 48Khz is more than enough. In my opinion, 96Khz is overkill and a waste of disk space and resources
Cj
This is not about recording - it's about the quality of sound produced by digital effects and synthesizers which is affected by aliasing. Let's not derail this topic into a conversation of whether or not this is true, and take on that aliasing is indeed very noticable. Feel free to start another thread about the question of validity of this statement, I'll gladly provide very good evidence to back up my stand. ORIGINAL: altima_boy_2001
As for the other point, can the best sample rate conversion algorithms work in real time on a CPU (ie. without hardware support)? I don't know so I'm asking...That would be a requirement, but I suppose it's possible if they have oversampler plugins available...
Even PPHS gives you incomparably less distortion than a single plugin in your project exhibiting aliasing. For the purpose of this topic the answer is 'yes'. However, to further reinforce this, even Sonar has Ozone resampling that is meant for individual clips, and can run this algorithm heavily multitracked, so as you can see it's not a big problem on the CPU side. ORIGINAL: pwal
you can oversample individual VSTs using something like VST Oversampler
Thank you - I already know about this - I actually beta tested it :) It requires upsampling and downsampling for every plugin that exhibits aliasing. This means that there are probably 60 such conversions happening in a typical project of mine. Therefore it has several problems: 1. If all plugins run at 2x the sample rate, whether or not they're oversampled one by one or as a whole by running the project at a double sampling rate, those 60 processes require a some, substantial or not, processing power. 2. 60 resampling operations take their toll on the quality of the sound. Even more so if you chain plugins and have the resultant errors accumulate. 3. The transport between subsequent plugins is still at the lower frequency. High frequency content is lost. The difference can be easily noted with the classic example of running something through a distortion and then a ring modulator. With a higher sample rate transport the ring modulator will have a much much richer output. 4. Maintaining loads of different plugins with the oversampler is just cumbersome. 5. It is sometimes unstable. Therefore a single resampling operation at the output is by far better than multiple resampling operations earlier on.
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John
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 15:39:39
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What VSTis produce higher sampled output then the project?
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dcastle
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 15:42:58
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However, at those settings, many my VSTi's really sound very bad because of the aliasing. How do you know it is because of 'aliasing'?
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CJaysMusic
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/12 16:15:06
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What VSTis produce higher sampled output then the project? How do you know it is because of 'aliasing'? I was going to ask, but lost interest. I know in my heart and my ears that going to 192khz is a gimmick and just a waste of disk space and resources. If the Poster wants to believe this advertising gimmick, oh well. Maybe he'll come around. Here is is problem.. as many people I am for the time being stuck with a card that only supports 44.1 or 48kHz audio rates On board or a sound blaster card. I'm sure its the quality of his converters and not the sample rates. 48khz is more than enough. But I'm not in the mood for arguing right now. Maybe later.. Cj
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cheater
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 03:33:54
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ORIGINAL: John
What VSTis produce higher sampled output then the project?
John, very many VST's that have internal oversampling. Notice I said VST, because channel plugins do the same. Take Concrete FX Predator as an example, it allows you to set oversampling to up to 16x. This means that if your project is running at 48 kHz, then Predator's engine is running at a 768 kHz sampling rate internally, and then gets sampled down to 48 kHz. Then I'll use an EQ, and that oversamples 2x. This means that it converts the signal from 48kHz to 96 kHz by oversampling, and then processes the sound, and then downsamples to 48 kHz again. Then I use, say, Voxengo Elephant, and that has oversampling too, so it oversamples 4x. This means it upsamples to 192 kHz, and then does its processing, and then downsamples to 48 kHz again. Then I'll use a gate, and that does 16x oversampling, so it'll oversample the signal to 768 kHz and then does the processing and then downsamples to 48 kHz. Now, most of those plugins allow you to turn off oversampling and run them at the project's sampling rate. If the project were 768 kHz back to back, there would be less oversampling and downsampling operations in the chain. Also, to give you an example, the gate is a ring modulator - so, specifically during its attack phase, it would provide a more accurate emulation of real-world equipment's behavior, since the high-frequency content is carried over from previous operations. We all know that a ring modulator has the perfect ability to take that supersonic content and frequency-shift it down into the audible realm via the sidebands it outputs. So no - the plugins don't output at a sample rate higher than the project because that's impossible technically from the point of view of the coder. However in many plugins nowadays everything up until the very end, when the plugin outputs, happens at astonishingly high sampling rates of between 3/4 MHz and 3 MHz. So the plugin 'almost' outputs at 3MHz, and certainly its sound quality benefits from the lack of aliasing distortion. ORIGINAL: dcastle
However, at those settings, many my VSTi's really sound very bad because of the aliasing.
How do you know it is because of 'aliasing'?
Sorry - I didn't express myself clearly. Of course there are multiple things that can make a DSP algorithm sound bad, however aliasing is a big culprit in many cases. If a compressor has a 'weird' attack that stands out and doesn't fit well with the signal, that's often aliasing (that's not to say that they would sound perfect otherwise, but aliasing can make things sound really weird). But most pathologically it's VST instruments that exhibit the nastiest aliasing. If you take FabFilter Twin, or even the newly released Twin2, and play the highest registers of some patches while running the project at 48 kHz, you might experience very bad aliasing. I'm not even going to mention old favourites like Pro-53. Actually if you start playing scales going up, at one point you will hear a pitch that is going down as you go up the scale. This is exactly aliasing. If you run the same plugin at 192 kHz the aliasing is gone and the notes stop sounding inharmonic. On the other hand, the already mentioned ConcreteFX Predator allows great experimentation with this: try running it at 44.1kHz with the oversampling in the 'advanced' panel turned all the way down. Turn off the oversampling filter, too. See if you can find aliasing the way I described: you might need to bypass all filters and set the note sustain to maximum. Then set the aliasing to 4x, and compare. The synth engine is now running at 176.4 kHz and will produce noticably less aliasing. Then set the aliasing to 1x but set the project to 176.4 kHz. The synth engine is running at 176.4 kHz again and should produce the exact same amount of aliasing, which is to say 'little to none'. (note: 176.4 kHz is not a popular sampling frequency but many sound cards allow it) ORIGINAL: CJaysMusic
I was going to ask, but lost interest. I know in my heart and my ears that going to 192khz is a gimmick and just a waste of disk space and resources. If the Poster wants to believe this advertising gimmick, oh well. Maybe he'll come around.
Gimmick? That's very interesting considering every single Sonar 8 user gladly gives into this gimmick praising TL-64 for its oversampling option, which means that inside it it processes audio at a supersonic sampling rate. Or maybe Rob Papen likes to support gimmicks as well with the Rob Papen Predator made by Concrete FX that processes audio at nearly 1 MHz. Or let's see what Bob Katz has to say about this in his book: ORIGINAL: Bob Katz
Double Sampling?
The most advanced digital equalizers and dynamics processors use double sampling technology, which means that the internal sampling rate is doubled to reduce aliasing distortion.
[...]
dynamics processors benefit because non-linear processing generates severe aliases of the sampling rate, and the higher the sample reate, the less aliasing.
Later, Bob goes on to compare a couple dsp limiters, Weiss DS1-MK2 and Waves L2. This is what he has to say: ORIGINAL: Bob Katz
Note the oversampling processor exhibits considerably lower quantization distortion. However, the switchable safety limiter of the Weiss, which is not oversampled, produces considerable alias distortion even at 1dB limiting. At 88.2 kHz and above, the Weiss safety limiter and the Waves perform measurably better[...]Thus there is considerable advantage of doing all our processing at higher rates. Note that this has been written in 2002. Nowadays dsp developers know that some algorithms have a certain way of accumulating aliasing which means that even if two dsp algorithms used on their own don't exhibit aliasing distortion, if the output of one is used in another, they will exhibit it together. This means both have to be upsampled for this purpose. This can accumulate with the amount of stages, so if there are 2 stages you might (in the simplest situation) need to oversample 2x, with 3 stages you might need to oversample 4x, while with 4 different algorithms you could need 8x. Nowadays most plugin developers oversample their plugins at 8x or 16x, so if you're running your project at a modest 48kHz you are processing audio at 768 kHz. Or maybe you've ever spoken to an owner of a Yamaha CS1x who mentioned to you that it has a very bad, dissonant, distorted sound in the highest registers - that's aliasing too. Aliasing of e.g. a virtual analog VST can be so bad that it is louder than the original note. Pro-53 is a typical example that exhibits this badly. It is so bad that you could create a preset I used to have that only plays the dissonant distortion in the sound! Usually with synthesizers that exhibit this problem the aliasing distortion is a pitch which is not harmonically related to your melody that is audible at 3 to 9 dB less than your original note playing, which means that it is very audible. It can be heard on the cheapest sound cards you can get right now, on 20-dollar "multimedia" speakers. I tried it and can attest empirical evidence of reproduction. Furthermore, I work on dsp algorithms myself, and knowing what aliasing is belongs to my responsibilities. Therefore I think your suggestion that my audio card is the culprit of the problem (not quoted in my post) is wrong.
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cheater
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 03:56:18
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If you look at a pretty typical sound chain of a VST synthesizer followed by a compressor, then a gate, you will note that each of those components - if you're using new plugins - can very likely work at 3/4 MHz internally. That's just a modest project sampling rate of 48 kHz, with the plugins internally oversampling at 16x. You can view this in this way: the sound chain is 48 kHz with the signal being temporarily upgraded to 3/4 MHz while it's being processed. However, that will not be representative of the truth, since exactly the processing is where the signal 'spends the most time'. So you might instead view the signal chain this way: the sound chain is 3/4 MHz with the sound being downgraded to 48 kHz between the processing stages. The only thing that happens after the sound is downgraded to 48 kHz in our compressor's outputs is that it gets upgraded by the gate's inputs. So that's a downgrade immediately followed by an upgrade again. There is no actual need for this downgrade to happen. In real world terms this can be compared to having a rack of Universal Audio and Manley and Neve processors, but connecting them together with something that downgrades, like a cheap cable. [1] Actually this downgrade and upgrade process creates further distortion since oversampling and downsampling algorithms inherently add distortion. Therefore it is much better to only have one such downsampling operation at the very output, instead of tens of instances of those algorithms all throughout the whole project. There's a higher chance than not that the output compressors you guys put on the master have internal oversampling anyways, therefore that single downsampling operation at the output happens anyways. The question is just to get rid of the other ones. But to reinforce what this topic is about, most older plugins don't have this inherent oversampling or it is something like 2x which is nowadays known not to be adequate in some scenarios. And many new ones, even from good developers, don't have this either. Therefore it is important to be able to run the project at high sampling rates, even if the output will be at 44.1 kHz.
[1] Now don't get me wrong, I'm not a big fan of cables made from oxygen free copper crystals pulled by one-eyed midgets in Peru during a full moon using a modified Czochralski process, but a properly thick copper cable terminated with wet solder points to good neutrik plugs is probably the minimum I'd approach such a rack with.
post edited by cheater - 2009/02/13 04:05:00
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Saintom
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 03:56:45
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ORIGINAL: cheater
ORIGINAL: John
What VSTis produce higher sampled output then the project?
John,
very many VST's that have internal oversampling. Notice I said VST, because channel plugins do the same. Take Concrete FX Predator as an example, it allows you to set oversampling to up to 16x. This means that if your project is running at 48 kHz, then Predator's engine is running at a 768 kHz sampling rate internally, and then gets sampled down to 48 kHz. Then I'll use an EQ, and that oversamples 2x. This means that it converts the signal from 48kHz to 96 kHz by oversampling, and then processes the sound, and then downsamples to 48 kHz again. Then I use, say, Voxengo Elephant, and that has oversampling too, so it oversamples 4x. This means it upsamples to 192 kHz, and then does its processing, and then downsamples to 48 kHz again. Then I'll use a gate, and that does 16x oversampling, so it'll oversample the signal to 768 kHz and then does the processing and then downsamples to 48 kHz.
Now, most of those plugins allow you to turn off oversampling and run them at the project's sampling rate. If the project were 768 kHz back to back, there would be less oversampling and downsampling operations in the chain. Also, to give you an example, the gate is a ring modulator - so, specifically during its attack phase, it would provide a more accurate emulation of real-world equipment's behavior, since the high-frequency content is carried over from previous operations. We all know that a ring modulator has the perfect ability to take that supersonic content and frequency-shift it down into the audible realm via the sidebands it outputs.
So no - the plugins don't output at a sample rate higher than the project because that's impossible technically from the point of view of the coder. However in many plugins nowadays everything up until the very end, when the plugin outputs, happens at astonishingly high sampling rates of between 3/4 MHz and 3 MHz. So the plugin 'almost' outputs at 3MHz, and certainly its sound quality benefits from the lack of aliasing distortion.
ORIGINAL: dcastle
However, at those settings, many my VSTi's really sound very bad because of the aliasing.
How do you know it is because of 'aliasing'?
Sorry - I didn't express myself clearly. Of course there are multiple things that can make a DSP algorithm sound bad, however aliasing is a big culprit in many cases. If a compressor has a 'weird' attack that stands out and doesn't fit well with the signal, that's often aliasing (that's not to say that they would sound perfect otherwise, but aliasing can make things sound really weird). But most pathologically it's VST instruments that exhibit the nastiest aliasing. If you take FabFilter Twin, or even the newly released Twin2, and play the highest registers of some patches while running the project at 48 kHz, you might experience very bad aliasing. I'm not even going to mention old favourites like Pro-53. Actually if you start playing scales going up, at one point you will hear a pitch that is going down as you go up the scale. This is exactly aliasing. If you run the same plugin at 192 kHz the aliasing is gone and the notes stop sounding inharmonic.
On the other hand, the already mentioned ConcreteFX Predator allows great experimentation with this: try running it at 44.1kHz with the oversampling in the 'advanced' panel turned all the way down. Turn off the oversampling filter, too. See if you can find aliasing the way I described: you might need to bypass all filters and set the note sustain to maximum.
Then set the aliasing to 4x, and compare. The synth engine is now running at 176.4 kHz and will produce noticably less aliasing.
Then set the aliasing to 1x but set the project to 176.4 kHz. The synth engine is running at 176.4 kHz again and should produce the exact same amount of aliasing, which is to say 'little to none'. (note: 176.4 kHz is not a popular sampling frequency but many sound cards allow it)
ORIGINAL: CJaysMusic
I was going to ask, but lost interest. I know in my heart and my ears that going to 192khz is a gimmick and just a waste of disk space and resources. If the Poster wants to believe this advertising gimmick, oh well. Maybe he'll come around.
Gimmick? That's very interesting considering every single Sonar 8 user gladly gives into this gimmick praising TL-64 for its oversampling option, which means that inside it it processes audio at a supersonic sampling rate.
Or maybe Rob Papen likes to support gimmicks as well with the Rob Papen Predator made by Concrete FX that processes audio at nearly 1 MHz.
Or let's see what Bob Katz has to say about this in his book:
ORIGINAL: Bob Katz
Double Sampling?
The most advanced digital equalizers and dynamics processors use double sampling technology, which means that the internal sampling rate is doubled to reduce aliasing distortion.
[...]
dynamics processors benefit because non-linear processing generates severe aliases of the sampling rate, and the higher the sample reate, the less aliasing.
Later, Bob goes on to compare a couple dsp limiters, Weiss DS1-MK2 and Waves L2. This is what he has to say:
ORIGINAL: Bob Katz
Note the oversampling processor exhibits considerably lower quantization distortion. However, the switchable safety limiter of the Weiss, which is not oversampled, produces considerable alias distortion even at 1dB limiting. At 88.2 kHz and above, the Weiss safety limiter and the Waves perform measurably better[...]Thus there is considerable advantage of doing all our processing at higher rates.
Note that this has been written in 2002. Nowadays dsp developers know that some algorithms have a certain way of accumulating aliasing which means that even if two dsp algorithms used on their own don't exhibit aliasing distortion, if the output of one is used in another, they will exhibit it together. This means both have to be upsampled for this purpose. This can accumulate with the amount of stages, so if there are 2 stages you might (in the simplest situation) need to oversample 2x, with 3 stages you might need to oversample 4x, while with 4 different algorithms you could need 8x. Nowadays most plugin developers oversample their plugins at 8x or 16x, so if you're running your project at a modest 48kHz you are processing audio at 768 kHz.
Or maybe you've ever spoken to an owner of a Yamaha CS1x who mentioned to you that it has a very bad, dissonant, distorted sound in the highest registers - that's aliasing too.
Aliasing of e.g. a virtual analog VST can be so bad that it is louder than the original note. Pro-53 is a typical example that exhibits this badly. It is so bad that you could create a preset I used to have that only plays the dissonant distortion in the sound! Usually with synthesizers that exhibit this problem the aliasing distortion is a pitch which is not harmonically related to your melody that is audible at 3 to 9 dB less than your original note playing, which means that it is very audible. It can be heard on the cheapest sound cards you can get right now, on 20-dollar "multimedia" speakers. I tried it and can attest empirical evidence of reproduction. Furthermore, I work on dsp algorithms myself, and knowing what aliasing is belongs to my responsibilities. Therefore I think your suggestion that my audio card is the culprit of the problem (not quoted in my post) is wrong.
You seem very knowledgeable, could you please tell me what sampling rate the last Bruce "sprinsting" (jk)record was recorded in? I'll even bet if Vanilla Ice recorded at 192, he would have better sounding music  ahh there is a question " if Ice Ice baby was recorded at 192, would it still be a hit?  Tom
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cheater
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 04:08:47
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ORIGINAL: Saintom
You seem very knowledgeable, could you please tell me what sampling rate the last Bruce "sprinsting" (jk)record was recorded in?
I'll even bet if Vanilla Ice recorded at 192, he would have better sounding music
Tom, ah, but again this is not about recording. Recording at 44.1 kHz is perfectly adequate and there is little more you could ask for. This is about *processing* digitally. I can bet you that both of those records, even if either was recorded digitally, was either: 1. processed with analogue processors with the digital recording only working as that - a digital multitrack - or 2. processed with high oversampling rates. I really think that the analogue route is more likely, but even if it was all digital: if you are talking about Springsteen's 'Magic Tour Highlights', then I can bet you that Bob Ludwig has read Katz' book, and has probably come up with some ideas on aliasing of his own, and therefore knows he should oversample his plugins really well :) [qote] ahh there is a question " if Ice Ice baby was recorded at 192, would it still be a hit? Tom Maybe they would have come up with a more original bassline thanks to that ;)
post edited by cheater - 2009/02/13 04:16:11
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dcastle
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 04:15:02
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Of course there are multiple things that can make a DSP algorithm sound bad, however aliasing is a big culprit in many cases. If a compressor has a 'weird' attack that stands out and doesn't fit well with the signal, that's often aliasing (that's not to say that they would sound perfect otherwise, but aliasing can make things sound really weird). But most pathologically it's VST instruments that exhibit the nastiest aliasing. Just saying it doesn't make it so! This is pseudo-science at its worst, or worse…
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tarsier
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 10:43:09
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ORIGINAL: cheater
Alternatively, does anyone of you cool guys know of a Windows application/driver (that's win xp I'm using) that would 'fake' support, so that it shows Sonar 192k compatibility, and therefore lets me select that option with Sonar? I guess such a driver would then, again, downsample the audio and output it to my usual ASIO or WDM driver.
I would ask the ASIO4ALL developer for this sort of support. That seems right up his alley. And see if you can get up to the DXD rate of 5.6 MHz. And to the naysayers: Those statements dismissing the OP's request are just the sort of statements that give "science" a bad rep. When people say that scientists are arrogant know-it-alls, this is exactly the sort of behavior they're talking about. What we have is a claim that higher sampling rates gives greater fidelity in processing (and by fidelity I mean audible fidelity). That is a testable scientific claim. The OP also gave a few references to those claims. Also subject to scientific validation (or not). Now, to my knowledge there has been no well designed study performed to test such a claim, and until then I can't say one way or another that the claim is pseudo-science. On the face of it, it might seem absurd. But there have been some good reasons given (like from Bob Katz) why it might not actually be absurd. Even simply looking at basic digital filter design shows that when processing close to the nyquist frequency you run into interesting problems that you don't run into when processing away from it. So if you want to boost at 18 kHz, you might actually get a better audible result when processing at 96 kHz (or higher) than at 44.1 kHz. You will most certainly get a different result, whether or not it's audible is subject to verification. I for one have never heard any benefits to processing at higher sampling rates. I do know that many plugins internally upsample (like the UAD ones) and many of those sound fantastic. Do they sound fantastic because they upsample? Do they sound fantastic because they are talented DSP designers and the upsampling is irrelevant? Does the upsampling simply make it easier to do quality DSP? Would all DSP processes benefit from upsampling? These are all very good questions I think, and until more good studies have been done, I don't think that anyone can answer yes or no to them in terms of audio quality. Frankly, I think the OP is on to something in terms of actually testing this stuff and I may just ask the ASIO4ALL guy for this myself.
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dcastle
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 11:06:15
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Greetings, ORIGINAL: tarsier
And to the naysayers: Those statements dismissing the OP's request are just the sort of statements that give "science" a bad rep. When people say that scientists are arrogant know-it-alls, this is exactly the sort of behavior they're talking about. What we have is a claim that higher sampling rates gives greater fidelity in processing (and by fidelity I mean audible fidelity). That is a testable scientific claim. Bad mouthing anyone who is not a true-believer is more religion than science. I find it interesting that you use "science" twice in this tirade along with "claim" twice — show me the science — prove the claims! But, please don't show me the same kind of claims that justify $30,000 cables. cheater came in here with an attitude and something to prove, which is not supported by what he has said. If he wants higher frequency plugins, then turn in a feature request. But, don't bother to bring up high-priced names in a boutique industry as "proof" of anything. That's how all kinds of weird, wacky, pseudo-science is marketed to a gullible public. Let's remember that cheater who is a true-believer of higher sample rates started off looking for... ORIGINAL: cheater
Alternatively, does anyone of you cool guys know of a Windows application/driver (that's win xp I'm using) that would 'fake' support, so that it shows Sonar 192k compatibility, and therefore lets me select that option with Sonar? I guess such a driver would then, again, downsample the audio and output it to my usual ASIO or WDM driver. It doesn't take a genius to figure out that he is getting worse distortion from his sound card than anything! When he upgrades to professional quality equipment and conducts true peer moderated double-blind tests, then… Respectfully, David
ASUS M3A78 AMD 9950 Quad 2.6G 8GB Shure • Rhode • Audio-Technica • Allen&Heath GL2200-24 MOTU 24i • Presonus Firepod • E-MU 1212m • Zoom H2 SONAR 2XL-8PE • Sound Forge 1-9 • Audacity 0.1-1.3
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tarsier
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 12:19:32
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ORIGINAL: dcastle
Bad mouthing anyone who is not a true-believer is more religion than science. I find it interesting that you use "science" twice in this tirade along with "claim" twice — show me the science — prove the claims!
Was saying "naysayer" harsh? I thought it was a fairly mild name. About in the ballpark as calling someone's writings a tirade. And I don't think you quite understood what I said--I specifically pointed out that the claims weren't proven. Please read what I wrote again. I'll try to say it in another way: There is a claim: digital signal processing can benefit audibly from upsampling. (note that I never said anything about whether it's true or whether I believe it) This is a claim that can be tested. There have been no tests to demonstrate one way or another whether it is "true" in the statistically significant sense. There are reasons to think this claim might be true: One of which is that many fantastic sounding plugins upsample internally--but do they sound fantastic because they upsample or some other reason. This is something that can be tested. The test hasn't been done.
But, please don't show me the same kind of claims that justify $30,000 cables.
No one brought up high-priced cables until you did. I suspect you did to try to group together the **** of high priced cables (the claim of which "they sound better" has been pretty well tested and refuted) with the untested claim of "upsampling sounds better". Please don't try to group the two together, since the one has been refuted while the other hasn't been properly tested. I also pointed out another reason upsampling might improve the audibility of DSP, the example of a process at 18 kHz. Reading back at what I wrote I think I did use "nyquist frequency" incorrectly, but I think the context was correct.
cheater came in here with an attitude and something to prove, which is not supported by what he has said.
Isn't that what science is? "I have something to prove, it isn't supported by any evidence..." and so then you figure out how to do a test to gather info about it. Gather evidence, one way or another. That's what cheater was asking for, a method to try it out. That's a worthy goal.
If he wants higher frequency plugins, then turn in a feature request. But, don't bother to bring up high-priced names in a boutique industry as "proof" of anything. That's how all kinds of weird, wacky, pseudo-science is marketed to a gullible public. Let's remember that cheater who is a true-believer of higher sample rates started off looking for...
Maybe cheater really is a true believer. But maybe if he runs the test he might find that he doesn't hear a difference. And given that no one has actually done a proper test, why would anyone shoot down a method that might bring a bit more data on the subject? Regarding name dropping: I didn't see it as "Bob Katz says it's so, therefore it's true". I read it as "Bob Katz has made some interesting points about upsampling. How can I try it out?" Maybe cheater took it as gospel. But he wanted to try it out. Perhaps you don't think it is worth trying, but I think it is. And nowhere in what I wrote do you find that I said anything close to "yes, upsampling is the way to go. It makes things sound so much better." I pointed out some untested claims, and asked some questions that could be tested. That's how you do science: you come up with an idea, figure out how to test it, tell other people about it and have them test it, figure out the ways in which your test was invalid, fix those ways, test it again, repeat until you're satisfied. My whole point was that 1) There have been reasons given to think that upsampling might improve the sound. 2) there haven't been any good double-blind tests done to test those reasons. 3) Therefore, you can't say that upsampling to improve the sound is **** or not.
It doesn't take a genius to figure out that he is getting worse distortion from his sound card than anything! When he upgrades to professional quality equipment and conducts true peer moderated double-blind tests, then…
Perhaps his hardware is crap. Perhaps he is on a wild goose chase. Perhaps if he upgrades to a stellar 44.1 kHz card then his VSTis will sound fantastic. But that's another scientific method example: My VSTis sound like crap. Is it because 1) I'm running at 44.1 kHz. 2) My soundcard is crap. 3) Some other reason. I think it's a great experiment to run, starting with the premise: Do the VSTis actually sound like crap? And you'd need to define: What does "sound like crap" mean? David, you're opposed to pseudo-scientific garbage like high priced cables. Me too, and I'm not convinced about most of "conventional wisdom" in the audio realm. Just like many people have a knee-jerk reaction to "double-blind study" and they dismiss what you have to say, I think you've had a knee jerk reaction to "high sample rate" and dismissed it out of hand. But in this case, I think you've missed an opportunity at teaching how science works and using the scientific method.
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John
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 13:39:45
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The problem to me is simple. Not one of the plugins mentioned output at a higher sample rate then the projects sample rate. Thus how is it possible to say what would sound better when its never been heard. This is not dealing with the supersonic issues that some swear by. This is simple in that the OP has stated that some plugins sound better with no down sampling. How would he know when he has never heard that ever! He attributes a loss of sonic quality due to aliasing that he has never been able to hear. It is not just the host that is causing the down sampling but the plugin itself. Then unless he has hardware that will not down sample his signal how will he hear it. Its a premise with no substance. It assumes things that can not be proven or even tested in any way that I know of.
I could be totally misunderstanding the OP but I don't think so.
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CJaysMusic
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 16:09:37
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I think this whole thread is funny, exspecailly when the poster has a card that only supports 44 and 48khz and makes false acusations on how he needs betters sound quality while using a sub par sound card....I think weve been had.
Cj
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John
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/13 16:35:29
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I think this whole thread is funny, exspecailly when the poster has a card that only supports 44 and 48khz and makes false acusations on how he needs betters sound quality while using a sub par sound card....I think weve been had.
Cj I think you are right on this! LOL. The whole thing is weird. The notion that he is loosing quality and knows it because he can prove it is very funny.
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/14 12:51:08
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Hey Everyone - The OP presents a reasonable discussion - let's stop derailing it with all the OT posts (which I removed). Best,
post edited by Willy Jones [Cakewalk] - 2009/02/14 12:53:07
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ChristopherM
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/14 13:21:27
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What VSTis produce higher sampled output then the project?
Sorry for the late entry, but don't Pentagon and z3ta+ offer oversampling facilities?
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RTGraham
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/14 14:26:22
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ORIGINAL: Willy Jones [Cakewalk]
The OP presents a reasonable discussion
I agree. While I question how necessary his request might be for most people in most environments, I don't see anything specifically false or unreasonable about what he's presenting. Having used samplers and sampling keyboards essentially since shortly after they were first introduced (Ensoniq Mirage, EPS, Emax, etc.), I'm *very* familiar with what classic aliasing sounds like, and the mathematical principles under which it comes into play. Anybody who came into recording, or at least digital audio, more recently, might not have pushed a sampler into aliasing, and might not be as familiar with the associated audio artifacts. It had not occurred to me that the same circumstances might occur within an audio processing plugin, but it certainly makes sense now that the original poster has brought it up and explained certain details. ORIGINAL: John
I think this whole thread is funny, exspecailly when the poster has a card that only supports 44 and 48khz and makes false acusations on how he needs betters sound quality while using a sub par sound card....I think weve been had.
Cj I think you are right on this! LOL. The whole thing is weird. The notion that he is loosing quality and knows it because he can prove it is very funny.
I disagree. As just about everyone here has either stated, argued, or agreed, the culprit shouldn't be 44.1k or 48k playback (as opposed to higher-sample-rate playback); and the original poster describes specific examples and details that lead *me*, at least, to believe that when he uses the word "aliasing" he knows what he's talking about. Is it possible that he's hearing something else? Sure. But is is also possible that everyone who says it's just his converters might be wrong themselves? Yep, that's possible too. ORIGINAL: dcastle
cheater came in here with an attitude and something to prove, which is not supported by what he has said. If he wants higher frequency plugins, then turn in a feature request. Are you kidding me? Have you gone back and re-read the first post in this thread (which, incidentally, is unedited)? I don't see an attitude *or* anything to prove. I see a simple, genuine, polite request for assistance from people whom the original poster clearly thought would be both knowledgeable and helpful. He only started "proving" things when he himself was challenged. And it's not specifically higher-frequency plugins that he's asking for help with. ORIGINAL: cheater
ORIGINAL: pwal
you can oversample individual VSTs using something like VST Oversampler
Thank you - I already know about this - I actually beta tested it :) It requires upsampling and downsampling for every plugin that exhibits aliasing. This means that there are probably 60 such conversions happening in a typical project of mine. Therefore it has several problems:
1. If all plugins run at 2x the sample rate, whether or not they're oversampled one by one or as a whole by running the project at a double sampling rate, those 60 processes require a some, substantial or not, processing power.
2. 60 resampling operations take their toll on the quality of the sound. Even more so if you chain plugins and have the resultant errors accumulate.
3. The transport between subsequent plugins is still at the lower frequency. High frequency content is lost. The difference can be easily noted with the classic example of running something through a distortion and then a ring modulator. With a higher sample rate transport the ring modulator will have a much much richer output.
4. Maintaining loads of different plugins with the oversampler is just cumbersome.
5. It is sometimes unstable.
Therefore a single resampling operation at the output is by far better than multiple resampling operations earlier on.
Now *here's* what I'd actually like to discuss, instead of having to defend the original poster against unnecessary and unreasonable attacks: Is a single resampling operation at the top and bottom of the chain really far better than multiple internal resampling operations at the plugin level? Wouldn't this require that *all* plugins in the project be capable of running, internally, at whatever oversampled clock rate the project is maintaining as its signal path? What if the project is globally oversampling to 768k, but there's a plugin in the project that maxes out at 96k? Won't there still potentially be aliasing issues (if aliasing is in fact what's happeining), depending on what frequencies are being processed and what the plugin is doing to them? And isn't there still the potential to have to do resampling going in and out of certain plugins? And how much more overhead would be added to an audio engine by having to run *everything* at a higher oversampled rate? It seems that just running the whole project in an oversampled environment might not solve all problems. Certainly an interesting point to discuss, though. Just my $0.02.
~~~~~~~~~~ Russell T. Graham Keys, Vocals, Songwriting, Production russell DOT graham AT rtgproductions DOT com www DOT myspace DOT com SLASH russelltgraham
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dreamkeeper
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/14 16:46:34
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ORIGINAL: RTGraham
Now *here's* what I'd actually like to discuss, instead of having to defend the original poster against unnecessary and unreasonable attacks:
Is a single resampling operation at the top and bottom of the chain really far better than multiple internal resampling operations at the plugin level? It depends I guess. It's mostly dynamics processors, tape/tube saturation, amp-sims, bit-crusher etc. that can benefit from higher samplerates - and of course synths with audio-rate modulation like FM or hard-sync. If many of those are used in a project, global oversampling could perhaps improve fidelity. And, like the OP points out, the supersonic content that one effect may generate at higher rate, could indeed affect the output of the next plug-in in the audible frequency ranges. Wouldn't this require that *all* plugins in the project be capable of running, internally, at whatever oversampled clock rate the project is maintaining as its signal path? What if the project is globally oversampling to 768k, but there's a plugin in the project that maxes out at 96k? That could indeed be a problem. Many algorithms, the most obvious being delays and reverbs, depend on "real-world" timing, yet are calculated from sample counts. Plug-ins therefore rely on the host to get the proper info about current samplerate, so they can adjust their calculations. Properly coded plug-ins should be able to deal with any reported samplerate, but the developers may have chosen to restrict the possible values to a "reasonable" range. Many include that in their specs, but not all do. Won't there still potentially be aliasing issues (if aliasing is in fact what's happeining), depending on what frequencies are being processed and what the plugin is doing to them? That can't be ruled out, yes. Actually, the more supersonic content is being passed, the higher that will raise the bar for the next effect ("96k recorders", read this again!). If downstream plug-ins add their own harmonic content to the already higher frequencies, chances are that even the oversampled rate won't be enough to prevent aliasing. And before you folks ask: aliased content can end up anywhere in the frequency range, not only near below Nyquist. And isn't there still the potential to have to do resampling going in and out of certain plugins? At least those that can't deal with the higher rates, yes. And how much more overhead would be added to an audio engine by having to run *everything* at a higher oversampled rate? For the kind of projects that probably more than 90% of users are running, it's most likely way too expensive. With a certain amount of critical plug-ins, a break-even could be reached, where piling on more of those would lead to less overhead with the global oversampling - not taking the possibly improved fidelity into account. Maybe a reasonable compromise would be an option to oversample individual FX-bins as a whole. As a next step one could imagine to dedicate certain user-definable signal paths, i.e. a chain (or tree) of buses to oversampling. Certainly enough possibilities to give developers some nightmares, hehe! werner PS: Oh, and Kudos to Willy Jones for calling the peanut gallery to order! 
"... must've been another of my dreams ..."
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cheater
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/14 18:57:51
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Hi guys,
Willy, thanks a lot for your post.
First I'd like to address some meta-conversation claims done here, to get them out of the way as quickly as possible. The '48 kHz sound card I'm stuck with' is not the end of the world for me, it's just the onboard audio in one of my home PC's. As one of my roles I'm a studio engineer, the last studio I worked with for a longer while was based completely on Protools and Digidesign hardware, with Yamaha main monitors. I can attest that aliasing is audible there as well, under certain conditions.
I can also say that I feel I have at least the minimum knowledge needed to be able to not act pseudoscientifically, having studied mathematics at university. The amount of rigor required to study such a subject completely rules out any possibility of such erroneous behavior from anyone attempting to study it at a university. You can rest assured that I am not acting upon novelty or pseudoscientific newspeak advertising.
Regarding edited posts: the edits were only spelling mistakes or *added* text, never removed. None of my edited posts had some sort of attacking captions inside them that were removed. I wish the forum allowed viewing previous versions of the posts, but it doesn't.
I don't wish to bad-mouth anyone and don't want to generate any sort of fuss.
I felt personally attacked with some claims and wanted to clear this up.
Now that this is done, let me get back to the topic.
There are some interesting points mentioned here:
1. Some plugins might benefit from higher sampling rates because of higher update rates
This is a point that many people forget about. Due to many reasons, plugin developers often decide to have internal signaling (not audio, but virtual 'control voltages' like envelopes or LFOs) at rates lower than audio rates. For example, some synthesizers might only have their envelope level recalculated once every 4 samples. This means that with a higher sampling rate, this synthesizer will allow snappier envelopes. However that's not even the beginning of this. Since a VCA is a ring modulator, and it's basically a multiply operation, what you have there is another problem: calculating multiplications of signals always generates aliasing. The question is whether that aliasing will be in the audible region (as it always is in 44.1k processing) or if it'll be in the supersonic region (with 96 kHz or more)
2. Some plugins might be unable to work at high sampling rates
Indeed, some plugins have precalculated information in them that's only calculated for popular sampling rates. However, if the plugin is written by a good coder, such information is all stored inside a single file and those definitions get reused all throughout the code. Therefore updating the plugin for new available sampling rates is usually a minor maintenance job and a recompile. In the worst case, if you really need to use a certain plugin, you can just opt out of using the higher sample rate. Please note however that my original problem was trying to run at 192k while my hardware only supports 48k. The 192 kHz sampling rate is hardly exotic. I expect almost all plugins to support it, and I expect all plugins to support 96k.
3. Some plugins don't need the oversampling
Indeed, there is no gain in oversampling when you have something like Sonalksis FreeG... which is just a volume fader.
4. Whether running everything at a high sampling rate is better than running just some things at high sampling rates
This is an issue to be decided on a case by case basis. Some people will find that they like it this way, some will not need it, some will deem it absolutely necessary. I'm probably in the first rather than the last group of people.
5. What are the tradeoffs of having repeated upsampling and downsampling
There's a point that upsampling and downsampling operations are computationally difficult. It depends: those operations are of varying quality. Pow-r 2 is a bit costy, but most plugins use PPHS or even linear resampling. I'm not experienced enough with dsp optimization to know whether the resampling in a simple filter/eq plugin takes a considerable amount of processor power, compared to the actual meat of dsp processing.
However the biggest problem with resampling is the phase smear it introduces into the sound passing through it. It's an accumulative error and it can really make the sound bad.
6. About aliasing generated by the inclusion of high-frequency content in a high sampling rate transport
This is a very interesting point and is not to be disparaged. This is only theoretic ground now, however two things are to be mentioned: 1. aliasing can be predicted by the developer and he can lowpass the output. This happens anyways, whether the output is downsampled or not. Low-passing is the natural stage that happens just before downsampling in any plugin. Therefore you can get a lot of supersonic frequency content without being burdened with aliasing. So: at the output of the plugin remove the downsampling while keeping the low pass filter, and you should be good.
Also, in case of trouble from a plugin, the skilled sound engineer will notice aliasing on the spectrogram and apply a low pass in the following fx bin.
7. About audibility of aliasing in synthesizers other than samplers
Indeed, synthesizers that are Virtual Analog can use wavetables (e.g. z3ta). Those wavetables are short samples - users of very great Ensoniq (hey! I just bought an esq80 by the way) synthesizers will recognize this as their legacy, a technology called 'allwaves' or 'transwaves' in Ensoniq terminology.
However, even algorithmic oscillators exhibit aliasing. But nowadays we have BLT techniques which make oscillators alias-free; however filters can still alias. Or the VCA. Or the overdrive stage. And as we see, not all techniques are being used even in very good plugins nowadays: go buy FabFilter Twin 2, released just recently, and you'll notice bad oscillator aliasing in high registers.
8. About audibility of aliasing in plugins other than synthesizers
This is a very interesting topic, especially because aliasing is much more difficult to spot in this kind of algorithm. Take compressors, for example. For what we want right now, they basically have overdrive stages. For example the moment of applying a very fast attack compression is a moment when overdrive happens. Overdrive distortion is very prone to aliasing: this aliasing can happen invariably of pitch of the input sound, and can have no audible relation to the original sound. Further, the aliasing can only happen at some points of the sound, and not during the whole length which is what you would get with sampler aliasing. The aliasing can only be as long as 3 samples long bursts of distortion. Or it can be even more subtle, exhibiting itself in the form of what can be described (somewhat imprecisely) as phase jitter. And again, update rates of internal 'Control Voltage' signals are higher with higher sampling rates.
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altima_boy_2001
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 02:15:07
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When you can pick up a portable 96kHz audio interface for as little as $199 (ex Echo AudioFire), about the cost of the Producer upgrade ($179) I don't know why you'd want a software solution. Heck, if you don't mind a PCI solution you can get a 192 kHz interface (ex EMU 1212M) for just $149. I'd prefer the Sonar developers to work on adding/updating other features right now myself. I think that the future engine-based oversampling fad might show up in Sonar 11 or 12 But by then decent 96/192kHz interfaces should be even more affordable so what's the point?  I do agree that implementing oversampling at the engine level would be the most accurate, efficient solution compared to separate plugin-based upsample/process/downsample.
post edited by altima_boy_2001 - 2009/02/16 02:24:55
You can use me as your eSoundz referral (altima_boy_2001).
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ChristopherM
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 04:11:29
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The amount of rigor required to study such a subject completely rules out any possibility of such erroneous behavior from anyone attempting to study it at a university. Rigorous? I am a high-court judge, therefore it is preposterous to suggest that it might have been me, whom you saw entering a brothel.
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Tom F
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 09:23:43
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aside from all that bashing and emotional stuff those "anti -op" guys stated:
i have an audiointerfave that supports 192khz...(well i wouldnt use it at such a high rate)
BUT:
i guess everyone can hear the difference of bass definition and genral fuller sound when vstis are used at 88 or 96
to say they sound same at 44 and 88 is just denying reality or deafnes - and the fact that some people here comment stuff they dont even read properly and therefore refere to the recording and not the internal sound generation...well..what should one think about that?
...trying to be polite... quick temper...trying to be...
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Geokauf
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 16:47:08
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it's about the quality of sound produced by digital effects and synthesizers which is affected by aliasing. Hello, I say this statement is patently untrue. And I suspect you have never actually heard "aliasing" to know what it would sound like. Even at 44.1KHz/16 bit you have about twice the fidelity and S/N as the great Studer/Neve, Studer/SSL combination in top world class studios that was used to make thousands and thousands of great albums. As back then, today, everyone is working with the same gear and at the same (dis)advantage. In other words, the bar is the same height for everyone. So how come this is a concern only to you (as I've never heard this issue broached before). I'd be interested in hearing examples of what you describe (as aliasing) rather than BS'g and beating about the bush. We're talking about the sound of audio and only the sound of audio can be used as an example. GK
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CJaysMusic
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 19:00:28
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In other words, the bar is the same height for everyone. So how come this is a concern only to you ( I was wondering that also. Nicely said..
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John
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RE: Using S8 with sample rates higher than the hardware supports? Oversampling the project
2009/02/16 19:35:09
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Hello,
I say this statement is patently untrue. And I suspect you have never actually heard "aliasing" to know what it would sound like. Even at 44.1KHz/16 bit you have about twice the fidelity and S/N as the great Studer/Neve, Studer/SSL combination in top world class studios that was used to make thousands and thousands of great albums. As back then, today, everyone is working with the same gear and at the same (dis)advantage. In other words, the bar is the same height for everyone. So how come this is a concern only to you (as I've never heard this issue broached before).
I'd be interested in hearing examples of what you describe (as aliasing) rather than BS'g and beating about the bush. We're talking about the sound of audio and only the sound of audio can be used as an example.
GK
This it precisely what I was asking too. I got flack for it. I do think that when some one purposes something that is rather new as a concept they must back it up with examples that all of us can deal with. Rhetoric alone is not going to pass muster here. Unless the OP can prove his allegations then its fantasy due to a misunderstanding of what is happening to audio in a plugin. The fact that CW got involved here is disappointing in that it has an effect of giving the notions expressed here some credence. All I ask is that the OP produce examples. Something I believe will be impossible.
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