Recording at 44 vs 192

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  • StuckfastStuckfast Frets: 384
    Again... up to a point, m'lud. The fly in the ointment with recording at 44.1 or 48 kHz is that all digital systems need filters on the input and output stages to eliminate frequencies above the Nyquist limit. Since there is no such thing as a filter with an infinite slope, the designers face a choice between using a gentle slope and potentially rolling off material within the audio range, or using a very steep slope and introducing ripples and phase shift into the response.

    So in fact the main technical argument for using higher sample rates is that it largely eliminates this compromise. You can have a nice gentle filter slope that doesn't introduce nasty artifacts, yet still stays out of the audio range.
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  • mr-macmr-mac Frets: 198
    ICBM said:
    Admittedly some years ago so the processing power available is different, but I did some comparisons in a recording studio when deciding what rate and depth to use for a project. I’m pretty sure they were using Cubase 5.

    Surprisingly, there was a noticable difference between 44.1KHz and 48, but none that we could hear between 48 and 96. Likewise there was a difference between 16 and 24-bit, but none between 24 and 32. So given that using higher quality slowed the system down drastically when mixing, we used 48KHz/24-bit.

    But that’s for recording, when you need more ‘data headroom’ for processing. On playback media, there’s no evidence that higher than 44.1/16 actually improves the quality, and it could actually make it worse... this is an old article but worth reading:

    https://xiph.org/~xiphmont/demo/neil-young.html

    (Don’t worry, it’s not by Neil Young ;).)
    Again depends on hardware but 48khz takes the processing sound further away from audio frequencies.  Most hardware can't resolve the advantage 192 might theoretically offer.  48/16 48/24 and 96/24 are where I'd be looking to use.

    there can be some advantage in efx applied in the daw software at the higher rates.  So there can be an advantage adding reverbs and other efx in 24/96 or 24/192 or higher but bring down to a 24/48 when creating final mixdown.

    agree it doesn't need to be high numbers but do think a lot of hardware produces better results at 48 than 44.1 but that's not because of the extra samples 
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  • mr-macmr-mac Frets: 198
    Most of problem with red book cd 44.1/16 is stupid levels of compression and sometimes a crap final mix down.

    high def doesn't solve the ridiculous levels of compression used.  
    There are plenty 44.1 and 48khz recording out there that prove the problem isn't inherent with that sample as they sound incredible.

    one thing to do if you capture and edit at 96 don't make final music avail at 44.1 make it 48.  If you plan on releasing a cd don't do all work at 96 or 192 etc etc but do it at 88.2 or 176.4.

    Why? We'll its very easy to accurately convert 176.4 or 88.2 to 44.1 with very little error.  But to change from 192 or 96 to 44.1 requires the software to interpolate the data (ie best guess of where wave would have been at point). 

    So if you wanna do a CD and HD use 88.2 or 176.4.... If doing video work use 48, 96 or 192.

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  • mr-macmr-mac Frets: 198
    edited June 11
    And whatever you do dont record or do the work in DSD even though some editors are available that are compatible now.  Almost every operation to change things or add efx requires the DSD data to be converted to multibit have work/change applied then converted back (ok happens in background but still two codec conversions just to do a basic operation on data isn't optimal and will degrade it).
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  • CirrusCirrus Frets: 3611
    edited June 11
    Stuckfast said:
    Again... up to a point, m'lud. The fly in the ointment with recording at 44.1 or 48 kHz is that all digital systems need filters on the input and output stages to eliminate frequencies above the Nyquist limit. Since there is no such thing as a filter with an infinite slope, the designers face a choice between using a gentle slope and potentially rolling off material within the audio range, or using a very steep slope and introducing ripples and phase shift into the response.

    So in fact the main technical argument for using higher sample rates is that it largely eliminates this compromise. You can have a nice gentle filter slope that doesn't introduce nasty artifacts, yet still stays out of the audio range.
    Absolutely - I wouldn't argue against that. So the real-world answer is that the 96kHz recording will be as much better than the 44.1kHz equivalent as the nyquist filters are audible to you. Personally, I can't really tell the difference on my RME Fireface. I think most converters are good enough in that regard these days to make that pretty much a non issue - not always the case, I remember the old Motu converters I used in 2003. You could hear what they were losing on playback.

    So, in practice, it comes down to how well the converters were engineered. Personally, I have no problem with my signal being, say, 1dB down and a few degrees out of phase by the highest frequency I could possibly hear. It pales in comparison to the things I'll do to the music by the time it's mixed down to stereo.  , and it pales in comparison to things like the quality of the clock, the op-amps on the way in and out...

    There's also the issue of aliasing and extra inter-modulation distortion if you record ultrasonic signals you don't actually need - record your project at 96k and you might have ultrasonic stuff that takes up headroom, interacts with signals in the audible range etc. At some point you down-sample the signal and you hope that stuff is going to be dealt with properly by the software.

    These are all very minor points, and my feeling is that the downside of capturing unnecessary info in the first place pretty much balances out the minor improvement in the nyquist filter situation.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • CirrusCirrus Frets: 3611
    mr-mac said:
    Most of problem with red book cd 44.1/16 is stupid levels of compression and sometimes a crap final mix down.

    high def doesn't solve the ridiculous levels of compression used.  
    There are plenty 44.1 and 48khz recording out there that prove the problem isn't inherent with that sample as they sound incredible.

    one thing to do if you capture and edit at 96 don't make final music avail at 44.1 make it 48.  If you plan on releasing a cd don't do all work at 96 or 192 etc etc but do it at 88.2 or 176.4.

    Why? We'll its very easy to accurately convert 176.4 or 88.2 to 44.1 with very little error.  But to change from 192 or 96 to 44.1 requires the software to interpolate the data (ie best guess of where wave would have been at point). 

    So if you wanna do a CD and HD use 88.2 or 176.4.... If doing video work use 48, 96 or 192.

     agree with the first half - 44.1/16 is enough to reproduce fantastic audio quality.

    Re' the sample rate conversion, that seems intuitively right but apparently it's not at all. Whether you're downsampling to 1/2 rate or some very complex ratio, the maths is sound. I've seen enough clever people say it that I'm inclined to believe them - including someone who designs A/D and D/A converters.
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • mr-macmr-mac Frets: 198
    Cirrus said:
    mr-mac said:
    Most of problem with red book cd 44.1/16 is stupid levels of compression and sometimes a crap final mix down.

    high def doesn't solve the ridiculous levels of compression used.  
    There are plenty 44.1 and 48khz recording out there that prove the problem isn't inherent with that sample as they sound incredible.

    one thing to do if you capture and edit at 96 don't make final music avail at 44.1 make it 48.  If you plan on releasing a cd don't do all work at 96 or 192 etc etc but do it at 88.2 or 176.4.

    Why? We'll its very easy to accurately convert 176.4 or 88.2 to 44.1 with very little error.  But to change from 192 or 96 to 44.1 requires the software to interpolate the data (ie best guess of where wave would have been at point). 

    So if you wanna do a CD and HD use 88.2 or 176.4.... If doing video work use 48, 96 or 192.

     agree with the first half - 44.1/16 is enough to reproduce fantastic audio quality.

    Re' the sample rate conversion, that seems intuitively right but apparently it's not at all. Whether you're downsampling to 1/2 rate or some very complex ratio, the maths is sound. I've seen enough clever people say it that I'm inclined to believe them - including someone who designs A/D and D/A converters.
    Ears tell me different. And a lot of very clever audio recording people do it that way.  Downsmspling  in a sympathetic rate (with clock dividable) takes zero guesswork or computer heavy lifting.  interpolating to a non sympethetic freq requires the software to do heavy calculations and interpolate date points.  I'm not saying difference is massive but it does bring in potential for not being close to 1:1.

    Things change though and all software is different so millage may vary.  Seems a no brainer to avoid a conversion stage that requires calculations to be made. 

    like i say on high end audio gear in past i have been able to tell which was interpolated and which wasn't..  Bit my ears now and software now may have got to stage its not really a big consideration.
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  • normula1normula1 Frets: 263
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
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  • mr-macmr-mac Frets: 198
    normula1 said:
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
    The difference heard could just be how well the dac handles each rate though rather than there being an audible difference.  The chip in dac could perform better at 192 than 48.
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  • StuckfastStuckfast Frets: 384
    It's pretty unlikely that there is any useful information coming from your turntable that can't be captured at 16-bit/44.1kHz. If I remember right the typical frequency response of a good vinyl master is about 30Hz - 15kHz and the dynamic range maybe 65dB at most.
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  • Winny_PoohWinny_Pooh Frets: 3073
    normula1 said:
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
    Conduct an ABY test, you may be surprised to realise that you are not hearing what you think you are.
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  • thegummythegummy Frets: 1131
    normula1 said:
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
    Conduct an ABY test, you may be surprised to realise that you are not hearing what you think you are.
    That's it, I'm surprised how often people trust their ability to hear such subtleties, it's so susceptible to placebo and other biases.
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  • Winny_PoohWinny_Pooh Frets: 3073
    thegummy said:
    normula1 said:
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
    Conduct an ABY test, you may be surprised to realise that you are not hearing what you think you are.
    That's it, I'm surprised how often people trust their ability to hear such subtleties, it's so susceptible to placebo and other biases.
    Its been theorized amongst some pros that many of the discrepancies listeners percieve are due to small differences in listening position rather than actual audible differences. (especially in relation to tweeters) 
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  • thegummythegummy Frets: 1131
    thegummy said:
    normula1 said:
    I do a fair bit of digitising Vinyl for use on my phone which has a HiRes DAC. My turntable also has a built in HiRes DAC which connects via USB and records at 192k/24bit. I then convert down to 48k/24bit and can definitely hear a difference between the two. However given the huge file size of the 192k recordings I tend to use the 48k ones.
    Conduct an ABY test, you may be surprised to realise that you are not hearing what you think you are.
    That's it, I'm surprised how often people trust their ability to hear such subtleties, it's so susceptible to placebo and other biases.
    Its been theorized amongst some pros that many of the discrepancies listeners percieve are due to small differences in listening position rather than actual audible differences. (especially in relation to tweeters) 
    Great point - that absolutely makes a difference among many many other things. For me discovering the truth about subtleties just allowed me to be happy in not bothering with tiny things like a tape sim for example - unless it's an exaggerated effect it's just going to make so little difference to the listener compared to so many other things.
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  • normula1normula1 Frets: 263
    My son and I did blind listening tests switching randomly between the recordings and invariably we could pick out the hires ones which was a bit of a surprise as I'm by no means a hifi cork sniffer.
    This will read like complete b*llocks but it was more the sense of space or I suppose openess rather than the music itself that was more pleasing to listen to.

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  • mr-macmr-mac Frets: 198
    No need for aby or ABX as non believers always sling at people who hear a difference.... A totally blind AB would work better if you get to hold the switch so can flick back and forth between the different files playing same music as and when you wish.

    My dac comes in 3 boxes one being a totally disconnected battery psu.  Totally blind on headphones i could pick within 3 flicks of switch if it was in battery or ac (even though a hifi reviewer reckoned the battery psu made very little difference.  it did abd was certainly noticeable and easy to pick).

    I reckon AB with ability to choose when it flicks is best test as you cab flick as often or as little as you like between the two.  Now pick best and repeat test a few times with A and B allocated randomly and see if pick same one over and over.

    it is also easy to hear it doesn't all sound he same by how people build systems.  Give two people same budget and tell em to buy a hifi.  One may come back with all one make and nice sounding system... The other comes back with auditioned separates which sounds night and day better than other system.  Well if we can't hear it and difference doesn't exist how come people can put together separates that sound much better than a random selection? Cos you can hear it.  Yes i am sure one day to next and mood, or moving head, or expectations can come in to it but differences also exist and can be heard too.

    Randomly allocated AB would seperste wheat from chaf.  ABX is flawed as its 3 options and your brain can hear one to next but can't then compare 1st to 3rd so easily.  And there is no need for the X when it really offers no advantag over a blind random AB with repeatable results .

    i remember i went to get an amp once and auditioned a pile of s/h hi-fi amps with high end atc monitor speakers.  Everything sounded good.  But the underdog we just chucked into mix as was in budget cleanly blew rest away.  My mate was with me and as soon as track we'd been using started we both looked at each other at same time and went holy shit.  It was a Sony ES amp which destroyed Marantz, quad a few others and even sounded better than a small krell integrated.  In my younger days when I could still heat bats I would happily have taken a blind AB test on hi-fi components and repeat and give same results.  Not sure if I'd be up for 192 v 48 these days though.  May try it at some point on good headphones and see if i can identify out a random a/b without being bothered which wins.
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  • mr-macmr-mac Frets: 198
    normula1 said:
    My son and I did blind listening tests switching randomly between the recordings and invariably we could pick out the hires ones which was a bit of a surprise as I'm by no means a hifi cork sniffer.
    This will read like complete b*llocks but it was more the sense of space or I suppose openess rather than the music itself that was more pleasing to listen to.


    again it could be your particular dac hardware just handles 192 better than 48 rather than the file itself being the issue 
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  • thegummythegummy Frets: 1131
    mr-mac said:
    No need for aby or ABX as non believers always sling at people who hear a difference.... A totally blind AB would work better if you get to hold the switch so can flick back and forth between the different files playing same music as and when you wish.

    My dac comes in 3 boxes one being a totally disconnected battery psu.  Totally blind on headphones i could pick within 3 flicks of switch if it was in battery or ac (even though a hifi reviewer reckoned the battery psu made very little difference.  it did abd was certainly noticeable and easy to pick).

    I reckon AB with ability to choose when it flicks is best test as you cab flick as often or as little as you like between the two.  Now pick best and repeat test a few times with A and B allocated randomly and see if pick same one over and over.

    it is also easy to hear it doesn't all sound he same by how people build systems.  Give two people same budget and tell em to buy a hifi.  One may come back with all one make and nice sounding system... The other comes back with auditioned separates which sounds night and day better than other system.  Well if we can't hear it and difference doesn't exist how come people can put together separates that sound much better than a random selection? Cos you can hear it.  Yes i am sure one day to next and mood, or moving head, or expectations can come in to it but differences also exist and can be heard too.

    Randomly allocated AB would seperste wheat from chaf.  ABX is flawed as its 3 options and your brain can hear one to next but can't then compare 1st to 3rd so easily.  And there is no need for the X when it really offers no advantag over a blind random AB with repeatable results .

    i remember i went to get an amp once and auditioned a pile of s/h hi-fi amps with high end atc monitor speakers.  Everything sounded good.  But the underdog we just chucked into mix as was in budget cleanly blew rest away.  My mate was with me and as soon as track we'd been using started we both looked at each other at same time and went holy shit.  It was a Sony ES amp which destroyed Marantz, quad a few others and even sounded better than a small krell integrated.  In my younger days when I could still heat bats I would happily have taken a blind AB test on hi-fi components and repeat and give same results.  Not sure if I'd be up for 192 v 48 these days though.  May try it at some point on good headphones and see if i can identify out a random a/b without being bothered which wins.
    The reason I think an abx is better than just ab is that it proves we are hearing a specific difference and can reliably tell when it's present or absent. When just deciding which we prefer out of 2 it is more abstract than specific.

    If anyone is wanting to do this, the media player foobar2000 comes with an abx tool built in.

    The only point of doing this is if you want to know the truth about what you can hear for yourself, it's pointless to prove anything to anyone else. Some people will believe things regardless of scientific proof as we all know.

    I'll just reiterate that doing this made a big positive impact on my productivity and enjoyment of mixing music and think it's worth doing for any mix decision you'd consider subtle.
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  • NelsonPNelsonP Frets: 771
    edited June 12
    This may be a noob question, but if you will eventually playback at 44.1khz then surely it's better to sample at this rate in the first place than at some other frequency that then needs to be rendered to 44.1.

    Any rendering would need to interpolate lots of data points and I can't see how it could do that as accurately as just sampling at 44.1khz (or a multiple of it) in the first place.

    Unless I'm missing something?
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  • thegummythegummy Frets: 1131
    NelsonP said:
    This may be a noob question, but if you will eventually playback at 44.1khz then surely it's better to sample at this rate in the first place than at some other frequency that then needs to be rendered to 44.1.

    Any rendering would need to interpolate lots of data points and I can't see how it could do that as accurately as just sampling at 44.1khz (or a multiple of it) in the first place.

    Unless I'm missing something?
    I think the advocates of higher sample rates are aiming for releasing as a download rather than CD.
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  • ToneControlToneControl Frets: 4487
    If you are recording mostly acoustic instruments, in a real space (i.e. with ambience rather than adding all the reverb later), it's worth it. Otherwise probably not
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  • Winny_PoohWinny_Pooh Frets: 3073
    this is fun if you havent tried it already. I have to admit that I liked one or two of the lower bitrate options on two tracks more than the higher bitrates. This was listening on my set of Neumann kh120s too. 

    https://www.npr.org/sections/therecord/2015/06/02/411473508/how-well-can-you-hear-audio-quality


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  • mr-macmr-mac Frets: 198
    edited June 13
    NelsonP said:
    This may be a noob question, but if you will eventually playback at 44.1khz then surely it's better to sample at this rate in the first place than at some other frequency that then needs to be rendered to 44.1.

    Any rendering would need to interpolate lots of data points and I can't see how it could do that as accurately as just sampling at 44.1khz (or a multiple of it) in the first place.

    Unless I'm missing something?
    effects work better applied at higher rates and cause less alaising.  So I'd always advocate capturing and working at a higher sympathetic sample rate 88.2 176.2 etc in case of wanting a 44.1 final product.
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  • mr-macmr-mac Frets: 198
    edited June 13
    And moving from a sample rate that can directly be divided to 44.1 as i suggest causes zero iterpolation.  It's only when moving from non dividable 48, 96, 192, 384 to 44.1 does the result need to be interpolated

    if final product is to be 48 then use 96, 192 etc
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  • mr-macmr-mac Frets: 198
    edited June 13
    thegummy said:
    mr-mac said:
    No need for aby or ABX as non believers always sling at people who hear a difference.... A totally blind AB would work better if you get to hold the switch so can flick back and forth between the different files playing same music as and when you wish.

    My dac comes in 3 boxes one being a totally disconnected battery psu.  Totally blind on headphones i could pick within 3 flicks of switch if it was in battery or ac (even though a hifi reviewer reckoned the battery psu made very little difference.  it did abd was certainly noticeable and easy to pick).

    I reckon AB with ability to choose when it flicks is best test as you cab flick as often or as little as you like between the two.  Now pick best and repeat test a few times with A and B allocated randomly and see if pick same one over and over.

    it is also easy to hear it doesn't all sound he same by how people build systems.  Give two people same budget and tell em to buy a hifi.  One may come back with all one make and nice sounding system... The other comes back with auditioned separates which sounds night and day better than other system.  Well if we can't hear it and difference doesn't exist how come people can put together separates that sound much better than a random selection? Cos you can hear it.  Yes i am sure one day to next and mood, or moving head, or expectations can come in to it but differences also exist and can be heard too.

    Randomly allocated AB would seperste wheat from chaf.  ABX is flawed as its 3 options and your brain can hear one to next but can't then compare 1st to 3rd so easily.  And there is no need for the X when it really offers no advantag over a blind random AB with repeatable results .

    i remember i went to get an amp once and auditioned a pile of s/h hi-fi amps with high end atc monitor speakers.  Everything sounded good.  But the underdog we just chucked into mix as was in budget cleanly blew rest away.  My mate was with me and as soon as track we'd been using started we both looked at each other at same time and went holy shit.  It was a Sony ES amp which destroyed Marantz, quad a few others and even sounded better than a small krell integrated.  In my younger days when I could still heat bats I would happily have taken a blind AB test on hi-fi components and repeat and give same results.  Not sure if I'd be up for 192 v 48 these days though.  May try it at some point on good headphones and see if i can identify out a random a/b without being bothered which wins.
    The reason I think an abx is better than just ab is that it proves we are hearing a specific difference and can reliably tell when it's present or absent. When just deciding which we prefer out of 2 it is more abstract than specific.

    If anyone is wanting to do this, the media player foobar2000 comes with an abx tool built in.

    The only point of doing this is if you want to know the truth about what you can hear for yourself, it's pointless to prove anything to anyone else. Some people will believe things regardless of scientific proof as we all know.

    I'll just reiterate that doing this made a big positive impact on my productivity and enjoyment of mixing music and think it's worth doing for any mix decision you'd consider subtle.
    Umm and how does repeatable results from a hidden random AB not? ABX if flawed to hear the smallest differences as only hear two of them back to back and brain doesn't work that way

    if i can pick same item as best time after time in a random hidden AB then it also proves there is a difference. 
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  • thegummythegummy Frets: 1131
    mr-mac said:
    thegummy said:
    mr-mac said:
    No need for aby or ABX as non believers always sling at people who hear a difference.... A totally blind AB would work better if you get to hold the switch so can flick back and forth between the different files playing same music as and when you wish.

    My dac comes in 3 boxes one being a totally disconnected battery psu.  Totally blind on headphones i could pick within 3 flicks of switch if it was in battery or ac (even though a hifi reviewer reckoned the battery psu made very little difference.  it did abd was certainly noticeable and easy to pick).

    I reckon AB with ability to choose when it flicks is best test as you cab flick as often or as little as you like between the two.  Now pick best and repeat test a few times with A and B allocated randomly and see if pick same one over and over.

    it is also easy to hear it doesn't all sound he same by how people build systems.  Give two people same budget and tell em to buy a hifi.  One may come back with all one make and nice sounding system... The other comes back with auditioned separates which sounds night and day better than other system.  Well if we can't hear it and difference doesn't exist how come people can put together separates that sound much better than a random selection? Cos you can hear it.  Yes i am sure one day to next and mood, or moving head, or expectations can come in to it but differences also exist and can be heard too.

    Randomly allocated AB would seperste wheat from chaf.  ABX is flawed as its 3 options and your brain can hear one to next but can't then compare 1st to 3rd so easily.  And there is no need for the X when it really offers no advantag over a blind random AB with repeatable results .

    i remember i went to get an amp once and auditioned a pile of s/h hi-fi amps with high end atc monitor speakers.  Everything sounded good.  But the underdog we just chucked into mix as was in budget cleanly blew rest away.  My mate was with me and as soon as track we'd been using started we both looked at each other at same time and went holy shit.  It was a Sony ES amp which destroyed Marantz, quad a few others and even sounded better than a small krell integrated.  In my younger days when I could still heat bats I would happily have taken a blind AB test on hi-fi components and repeat and give same results.  Not sure if I'd be up for 192 v 48 these days though.  May try it at some point on good headphones and see if i can identify out a random a/b without being bothered which wins.
    The reason I think an abx is better than just ab is that it proves we are hearing a specific difference and can reliably tell when it's present or absent. When just deciding which we prefer out of 2 it is more abstract than specific.

    If anyone is wanting to do this, the media player foobar2000 comes with an abx tool built in.

    The only point of doing this is if you want to know the truth about what you can hear for yourself, it's pointless to prove anything to anyone else. Some people will believe things regardless of scientific proof as we all know.

    I'll just reiterate that doing this made a big positive impact on my productivity and enjoyment of mixing music and think it's worth doing for any mix decision you'd consider subtle.
    Umm and how does repeatable results from a hidden random AB not? ABX if flawed to hear the smallest differences as only hear two of them back to back and brain doesn't work that way

    if i can pick same item as best time after time in a random hidden AB then it also proves there is a difference. 
    Don't know what you mean in the first paragraph. If you listen to a then x back to back then you can compare them just as easily whether you know it's going to be different or not.

    I don't think it could be possible to compare ab blind with the same 2 pieces of audio a load of times and pick the same one every time but then fail an abx.

    If it's actually known in science that this is possible please do link any articles that explain it.
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  • flying_pieflying_pie Frets: 434
    thegummy said:

    I don't think it could be possible to compare ab blind with the same 2 pieces of audio a load of times and pick the same one every time but then fail an abx.

    If it's actually known in science that this is possible please do link any articles that explain it.
    There is a perfect scientific explanation as to how that is possible:

    Chance. 
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  • thegummythegummy Frets: 1131
    Been thinking about this while getting ready for work. If someone used software to blind AB test 2 pieces of audio (at least a dozen times) and chose that they preferred the same one at least 90% of the time then it would prove there must be an audible difference for it to be consistent. That person would definitely also pass an ABX test with the same 2 pieces.

    What could happen, though, is that someone could pass an ABX test and hear a subtle difference when comparing 3 files where 2 of them are different to the other but then in a blind AB test of preference, not necessarily prefer the same one each time if they have no reference and are just listening the the audio as a whole. That's probably why ABX is used as the standard - a quick Google didn't really suggest AB is considered as an alternative to ABX.

    The process itself would be much different for me - listening through the AB and choosing which one I preferred would be quite abstract and only at the end, the results may prove there was a definite difference causing me to prefer the same one each time. With the ABX, I'd know each time if I was confidently hearing which was X or if I was just guessing. Of course, the results would be all that mattered as the confidence could have been wrongly felt but if I knew I was guessing I'd know at the time.

    If there is software you know of that does the blind AB test I'd appreciate a link. Would be interesting to see if I could pass an ABX on something very subtle like 320kpbs MP3 vs 192kbps MP3 but then fail the AB version.

    Might be a fantasy thinking I'll actually have time to do this to be honest but I'd appreciate the link none the less.
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  • CirrusCirrus Frets: 3611
    edited June 13
    mr-mac said:
    And moving from a sample rate that can directly be divided to 44.1 as i suggest causes zero iterpolation.  It's only when moving from non dividable 48, 96, 192, 384 to 44.1 does the result need to be interpolated

    if final product is to be 48 then use 96, 192 etc
    So, I know I'm going to look a bit like I'm trying to start a fight over some insanely geeky stuff, but I still think this is wrong. Just imagine we're having an argument with a smile on our faces over a pint in a pub, and also bare in mind (it it bare or bear?) that I'm totally happy to accept that you can hear the difference.

    It's been years since I looked into it but I've been doing a big of scouting around, trying to find old forums and articles.

    My understanding is that you'd be right, if converting to half the sample rate just involved dropping every other sample point. Then you can be happy, because the samples you're left with are all real samples that were actually measured by the converter as you recorded them.

    But it's not that simple. If you did that, it would be effectively like you recorded at the sample rate you're converting to, but had your nyquist filter set twice as high as it should be; have I explained that well enough? You've let signals into the digital domain that you're now not going to be able to reproduce because you don't have enough sample points any more - you've thrown half of them away. And in doing so you're not just throwing away the top octave of your bandwidth, as those higher frequency signals will still be present, modulating the samples you have kept hold of and producing aliasing.

    So, no matter what ratio of downsample conversion you do, the process isn't that simple because you still need to first brick wall filter the signal before the downsampling actually occurs. Now every sample point has been moved in amplitude. And that filtering process is far less transparent than interpolation of the signal's sample points. In fact, because filtering introduces phase shifts... your audio signal has been moved in time. Now, by the logic of avoiding interpolation, you're in hot water. Because *every* sample is now shifted relative to the audio signal* by different amounts for each frequency, by some tiny amount through the entire audio range! So you're not left with the actual sample values recorded, you're left with an entirely interpolated signal since the wave it's representing isn't the same any more.

    But that doesn't matter. because once you're in the box, once you're safely in the world of pure computation and you don't need to worry about the real-world practical side of digital sampling - analogue filters, quality of op-amps (distortion, slew rate), keeping power supply noise out the audio path, clocking errors - sampling theory is *perfect*. It's as certain as 1+1=2.

    So, the computer can calculate interpolated sample points when converting to a non-dividable target rate to an insane accuracy. It's got the whole signal perfectly recorded, and it can calculate any point along the theoretical band-limited analogue wave than the sample points would generate up to whatever precision you use to do the calculation. I'm not a maths or a programming guy, but I wonder how many decimal places you have to calculate to before any error is reduced below the ability of a human ear to pick out. And I wonder how many decimal places beyond that modern SRC processes go. I'd be interested to find out.

    Apparently in the olden days of digital sample rate conversion was a different process so simpler conversions used to be better, but these days it just doesn't matter. Personally, I'm happy that any error introduced will be so utterly insignificant that it can as well be discounted.

    It's also worth pointing out that as a mixer I'm mangling the audio in all kinds of ways - plugins do their own up and downsampling using different methods, the mix engine of my DAW moves audio about however it's been programmed to, dither may or may not be applied when it should, bit depths are constantly changing... there's so much going on.

    Now, on the subject of higher sample rates, this guy has made some fantastic plugs that I love;

    https://www.gearslutz.com/board/mastering-forum/968641-some-thoughts-quot-high-resolution-quot-audio-processing.html

    And he makes the point that by recorded a wider band than you want to end up with, you can end up introducing more distortion in the final product than if you'd ignored those higher frequencies from the off.

    Frankly, I think the whole world of digital sampling is fascinating and I'd *love* to understand it better.




    *or rather, the audio signal has been shifted relative to the sample points, which obviously don't move. But it's all relative!
    Captain Horizon (my old band);
    Very (!) Occasional Blog
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  • thegummythegummy Frets: 1131
    thegummy said:

    I don't think it could be possible to compare ab blind with the same 2 pieces of audio a load of times and pick the same one every time but then fail an abx.

    If it's actually known in science that this is possible please do link any articles that explain it.
    There is a perfect scientific explanation as to how that is possible:

    Chance. 
    Well there is a chance you could run a random number generator and the resulting binary code it generates could be a perfect mp4 encoding of the video of your great great grandfather's birth, it's all just down to probability lol

    So the tests would have to be done enough times to rule out chance as realistic. I think a surprisingly low number of tests would be needed before the probability of luck can be seen as very unlikely.
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