Ahh, I'm sorted then.pj geerlings wrote: it is possible to provide a nearly perfect rendition of an extremely boring tune - even at 44.1K sampling rate
Any use of going beyond 44100?
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- Boss Lovin' DR
- 14312 posts since 15 Mar, 2002 from the grimness of yorkshire
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- KVRist
- 487 posts since 19 Jan, 2003
The only point I was trying to get across in my reply, and admittedly I did a piss poor job of it, is that "information being lost" in not the issue. 44.1KHz is capable of storing the required information. The benefit from higher sampling rates is achieved because it makes the converter's job easier, for a variety of reasons that have been explained by people more knowledgable than I in this thread.ericj23 wrote: what i am convinced about is that your dac plays join the dots - it may check a few dots ahead to make sure it is not producing nonsense but if there arent a whole lot of dots (ie at high frequencies ) it loses information
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- KVRAF
- 2609 posts since 26 Aug, 2002 from here
as i stated in my last post the sacds i own are of cd's i already own - and with the miles ones i can confirm that the cd layer is exactly the same as the regular cd issue (they were mastered at the sametime)
i refuse to beleive that sony would deliberately hurt their sales of a cd (which in the case of kind of blue still sells well) to make sacd sound better
ps can your 44.1 system get the amplitude of the sine wave at even 10khz correctly - not unless the wave is in phase it can't - lost information
i refuse to beleive that sony would deliberately hurt their sales of a cd (which in the case of kind of blue still sells well) to make sacd sound better
ps can your 44.1 system get the amplitude of the sine wave at even 10khz correctly - not unless the wave is in phase it can't - lost information
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- KVRAF
- 4738 posts since 20 Feb, 2004 from Gothenburg, Sweden
yes it can, no information lost.ericj23 wrote:ps can your 44.1 system get the amplitude of the sine wave at even 10khz correctly - not unless the wave is in phase it can't - lost information
Stefan H Singer
https://dropshotaudio.com/
https://dropshotaudio.com/
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- KVRist
- 339 posts since 9 May, 2001 from Greece
I think Christian pretty much nailed it.
I keep hearing the sine wave arguments so very often. You cannot tell a sine wave from a square or sawtooth apart at high frequencies. You just can't. Timbre gets its properties from the structure of harmonics, a sawtooth at 15kHz will have the first harmonic at 30kHz, you cannot hear that. It will always sound like a sine wave to you.
Overall I think it boils down to this... if you're a musician, 44.1k or 48k is great. If you are an engineer/producer then sure, go for 96k. Besides, it makes for a cool cover label "Utilises superior 96kHz technology". 96kHz also sounds -subconsiously- more sexy... just invert the numbers.
The big problem is, that many beginners and ill-informed read this stuff. One might believe that doubling the sample rate will magically transform their music into a sublime piece of listening nirvana. Yes, I have talked to such people!
I'll take great music in mp3 or old-skool tracker format ANY DAY over any indifferent pieces of work that are followed by astronomical technical specifications.
I keep hearing the sine wave arguments so very often. You cannot tell a sine wave from a square or sawtooth apart at high frequencies. You just can't. Timbre gets its properties from the structure of harmonics, a sawtooth at 15kHz will have the first harmonic at 30kHz, you cannot hear that. It will always sound like a sine wave to you.
Overall I think it boils down to this... if you're a musician, 44.1k or 48k is great. If you are an engineer/producer then sure, go for 96k. Besides, it makes for a cool cover label "Utilises superior 96kHz technology". 96kHz also sounds -subconsiously- more sexy... just invert the numbers.
The big problem is, that many beginners and ill-informed read this stuff. One might believe that doubling the sample rate will magically transform their music into a sublime piece of listening nirvana. Yes, I have talked to such people!
I'll take great music in mp3 or old-skool tracker format ANY DAY over any indifferent pieces of work that are followed by astronomical technical specifications.
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Christian Schüler Christian Schüler https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=48996
- KVRist
- 266 posts since 23 Nov, 2004 from Hamburg, Germany
stefancrs wrote:yes it can, no information lost.ericj23 wrote:ps can your 44.1 system get the amplitude of the sine wave at even 10khz correctly - not unless the wave is in phase it can't - lost information
Look you here:

The upper graph is the continuous wave from the sequence 0,1,0,-1,etc. The lower graph is the wave from the sequence 0.71, 0.71, -0.71, -0.71, etc, which is the same as above, shifted by 45 degrees. (Where 0.71 is just short for the square root of 1/2, actually).
The point is that DAC conversion will just get everything right, provided is does a good interpolation and filtering. From what I can guess, CoolEdit considers 30 or so neighboring samples to calculate a point on the wave. Your high end DAC should likewise.
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- KVRist
- 109 posts since 15 Sep, 2004 from New Jersey
you people are smart. sample rate shmample rate. i just make music. hehe..,i need to take a music theory class or something I have no idea what half this stuff means
http://www.myspace.com/nickperez
^Music^
^Music^
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- KVRist
- 223 posts since 2 Aug, 2004 from Boston, MA
I don't know much about this, but I can give a specifc example. I have found that Reaktor instruments almost always sound better at higher sampling rates. I generally bump up the sampling rate to 88.1 within Reaktor, even if the project sampling rate is lower; I downsample when I render to audio, or later if I'm running the host at a higher sampling rate. Specifically, I've noticed on Steampipe, the physical modeling synth, that the tuning is less accurate (flatter) at lower sampling rates, espeically, for some reason, when I've closed the project and opened it again later. The tuning is corrected when I bump up the sampling rate. For whatever reason, after Reaktor parts are rendered and downsampled to 44.1, they still sound better than if they start out at 44.1 or whatever.
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fishbowl.tucson.az fishbowl.tucson.az https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=45536
- KVRist
- 415 posts since 23 Oct, 2004
I cannot make much of an argument for 96kHz, but I definitely prefer using 48kHz and not 44.1. I also prefer to use 24 bit instead of 16 bit, because I like to have a lot of dynamic headroom. If you are keeping everything in the ballpark of 0dB it doesn't really matter, since you're throwing it away.
But as for samplerate, I prefer to use 48kHz, because I know it's much easier for the hardware to deal with the integer sample rate, and also because I have outboard gear that has a very accurate word clock, and locks my recording system at 48kHz. So it's very, very convenient to work at 24/48 than anything else.
As for the comments that "it all goes to 16/44.1 eventually", well, that's only if you're making audio CD's with it. That's only a problem if you're going to production. I see no reason to do that. Keep your master at 24/48 and if you're encoding mpegs or oggs, go straight from that.
I agree that the argument is a tempest in a teapot, but I also believe that headroom is good, oversampling is good, and the accuracy of a good 48kHz clock is good. I don't think it's been all that long since I was using 1/4" tape and whatever gear I could get my hands on, so as far as I'm concerned, it's *all* good.
But as for samplerate, I prefer to use 48kHz, because I know it's much easier for the hardware to deal with the integer sample rate, and also because I have outboard gear that has a very accurate word clock, and locks my recording system at 48kHz. So it's very, very convenient to work at 24/48 than anything else.
As for the comments that "it all goes to 16/44.1 eventually", well, that's only if you're making audio CD's with it. That's only a problem if you're going to production. I see no reason to do that. Keep your master at 24/48 and if you're encoding mpegs or oggs, go straight from that.
I agree that the argument is a tempest in a teapot, but I also believe that headroom is good, oversampling is good, and the accuracy of a good 48kHz clock is good. I don't think it's been all that long since I was using 1/4" tape and whatever gear I could get my hands on, so as far as I'm concerned, it's *all* good.
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- KVRAF
- 4738 posts since 20 Feb, 2004 from Gothenburg, Sweden
That's due to implementation "flaws" made by the programmers, not due to what can be done within a specific sample rate. It's much easier making for instance aliasing oscillators than bandwidth limited oscillators, but the first will always give faulty results if you want for instance a clean sawtooth. If you want aliasing noise, it will give you that thoughcobalt wrote:I don't know much about this, but I can give a specifc example. I have found that Reaktor instruments almost always sound better at higher sampling rates. I generally bump up the sampling rate to 88.1 within Reaktor, even if the project sampling rate is lower; I downsample when I render to audio, or later if I'm running the host at a higher sampling rate. Specifically, I've noticed on Steampipe, the physical modeling synth, that the tuning is less accurate (flatter) at lower sampling rates, espeically, for some reason, when I've closed the project and opened it again later. The tuning is corrected when I bump up the sampling rate. For whatever reason, after Reaktor parts are rendered and downsampled to 44.1, they still sound better than if they start out at 44.1 or whatever.
-c
Stefan H Singer
https://dropshotaudio.com/
https://dropshotaudio.com/
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Stupid American Pig Stupid American Pig https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=4753
- KVRAF
- 7065 posts since 25 Nov, 2002 from not sure
If I ever get a record made- I will record 2 versions. One will be done at 44.1, the other will be at 96. I will send an equal amount of copies of each to the stores.
oh wait you are right, it is pointless, because most of the copies will get ripped to MP3, and that would ruin either recording. Maybe the people who hear it on the radio could help- oh wait, there is a 15 khz cutoff for FM. hmm well maybe xm will do it, aww damn, they compress the music as well. maybe at a club they will be able to tell when the music is being pumped at 115 db through 2 way speakers with 15" woofers and Horn tweeters, hmm maybe not. I know who can tell, the middle aged audiophile guy with the vintage Mcintosh tube amps, b&w speakers, and $500/foot cables. Damn, that guy doesnt listen to my music.
anyway, all I am saying is, sure try to get the best sound you can, but in the end it will most likely be played in less than ideal conditions...
oh wait you are right, it is pointless, because most of the copies will get ripped to MP3, and that would ruin either recording. Maybe the people who hear it on the radio could help- oh wait, there is a 15 khz cutoff for FM. hmm well maybe xm will do it, aww damn, they compress the music as well. maybe at a club they will be able to tell when the music is being pumped at 115 db through 2 way speakers with 15" woofers and Horn tweeters, hmm maybe not. I know who can tell, the middle aged audiophile guy with the vintage Mcintosh tube amps, b&w speakers, and $500/foot cables. Damn, that guy doesnt listen to my music.
anyway, all I am saying is, sure try to get the best sound you can, but in the end it will most likely be played in less than ideal conditions...
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- KVRAF
- 13446 posts since 14 Nov, 2000 from Hannover / Germany
What he said...S_A_P wrote: anyway, all I am saying is, sure try to get the best sound you can, but in the end it will most likely be played in less than ideal conditions...
There are 3 kinds of people:
Those who can do maths and those who can't.
Those who can do maths and those who can't.
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- Banned
- 4026 posts since 27 Jan, 2004
Well, S_A_P...It's very easy to justify using the highest possible quality and the reasons NOT to do it (as those you stated) are true but not valid...Why compromise your sound if you can get it to sound better?
Seems loosley retarded imo...And isn't better sound what we all want??
Ask any pro engineer if they care wether the peeps can tell the dif or not...It's not the point, the point is to sound your best and to sound as natural as possible!
And for the record, I hear vast improvements when the samplerate is above 44.1khz, and I'm hardly using a pro setup...only a modest home studio!
The point??
Use the highest quality you can and make sure you know how to transfer it to CD or other medium without making it sound crap...Othervise, stick to playing the banjo in your back garden
Seems loosley retarded imo...And isn't better sound what we all want??
Ask any pro engineer if they care wether the peeps can tell the dif or not...It's not the point, the point is to sound your best and to sound as natural as possible!
And for the record, I hear vast improvements when the samplerate is above 44.1khz, and I'm hardly using a pro setup...only a modest home studio!
The point??
Use the highest quality you can and make sure you know how to transfer it to CD or other medium without making it sound crap...Othervise, stick to playing the banjo in your back garden
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- KVRAF
- 4738 posts since 20 Feb, 2004 from Gothenburg, Sweden
Which ofcourse is due to the quality of your synths/effects/converters, not the sample rate itselffractalism wrote: And for the record, I hear vast improvements when the samplerate is above 44.1khz, and I'm hardly using a pro setup...only a modest home studio!
Bottom line is: atleast 48KHz (wider pass band than with 44, read earlier posts by pj) is enough as a "global" sample rate. some effects and sound generators need oversampling due to their innerworkings. if the developers making these would behave, they would give you the option to use internal oversampling instead of "forcing" the user to oversample the whole project and hence loosing a lot of cpu power.
Stefan H Singer
https://dropshotaudio.com/
https://dropshotaudio.com/
