Samplerates in max

Oct 18, 2007 at 10:35pm

Samplerates in max

Hi everyone,

I’m trying to listen to 3 different versions of an audio file (originally mixed at 96k/24), which are at 96/24, 48/24 and 44.1/16. I’m trying to see if there is a noticeable difference between the different conversions, but I need to make sure that I am listening to a certain file at te correct sample rate.

I know that I can set the samplerate using the ‘adstatus sr’ object (I can see my Digi 192′s sample rate change), and can use sfplay~ to play each audio file, but I am a bit confused as to at what sample rate sfplay will play each of the audio files.

I guess what I’m asking is, is there are way to play the 44.1 file at 44.1, 48k file at 48k and 96k file at 96k all in the same patch( not all playing at the same time, but the ability to switch between the different files), so that I can switch between and compare the different audio rates.

I do hope this makes sence; I have spent quite a long time trying to get this to work, but there doesn’t seem to be any difference in the listening tests – I am confident it is not down to bad equipment – am using Digi 192′s and Genelec monitors – so am thinking I havn’t made a correct max patch.

Thank you

Stuart

#34234
Oct 19, 2007 at 4:28am

I don’t know why you would listen to an ‘upsampled’ audio file?
there difference that you will notice is the varying effect on your cpu’s usage. you need to compare the same audio recorded at different rates, playing a 44.1 file back at 96 won’t do anything, the additional data just isn’t there.

benjamin

#115152
Oct 19, 2007 at 1:33pm

#115153
Oct 19, 2007 at 10:13pm

#115154
Oct 28, 2007 at 6:43am

Stuart Binns schrieb:
> Hi everyone,
>
> I’m trying to listen to 3 different versions of an audio file
> (originally mixed at 96k/24), which are at 96/24, 48/24 and 44.1/16.
> I’m trying to see if there is a noticeable difference between the
> different conversions, but I need to make sure that I am listening to
> a certain file at te correct sample rate.

Max will always convert it again to fit the set sampling rate. So choose
the driver with the sampling rate you want to test or set the sampling
rate (depends on your interface)…

Don’t expect any audible differences, the most crucial part is 1.
recording, and 2. processing. Higher sampling rates will have much less
aliasing, and smoother analog filters on the input. Thats it.

Sample rate conversion is a science in itself, but nowadays done usually
quite well…

If you did the sample rate conversion in Protools it should be fine…

Stefan


Stefan Tiedje————x——-
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#115155
Oct 29, 2007 at 2:05pm

I did the sample conversion in Pro Tools, and on conducting the test, I could notice a very small difference between the 44.1 and 48k files – the 48k seemed to be less ‘harsh’ in the middle frequencies; the 48k seemed to me to be a little bit smoother.

Thanks for all your input everyone – i really appreciate it.

#115156
Oct 29, 2007 at 2:30pm

What were you downsampling *from*? ISTM that you’d be more assured of
good results going 96->48 / 88.2->44.1 than, say, 96->44.1.

I saw a page a few months ago where someone had attempted to measure the
spectral artefacts left by SRC in various bits of software – it was
surprising how variable things were. The upshot seemed to be that Peak
and CoolEdit Pro / Audition did the best job.


Owen

Stuart Binns wrote:
>
> I did the sample conversion in Pro Tools, and on conducting the test,
> I could notice a very small difference between the 44.1 and 48k files
> – the 48k seemed to be less ‘harsh’ in the middle frequencies; the
> 48k seemed to me to be a little bit smoother.
>
> Thanks for all your input everyone – i really appreciate it.
>
>
>

#115157
Oct 29, 2007 at 4:44pm

>> I do hope this makes sence; I have spent quite a long time trying
>> to get this to work, but there doesn’t seem to be any difference
>> in the listening tests – I am confident it is not down to bad
>> equipment – am using Digi 192′s and Genelec monitors – so am
>> thinking I havn’t made a correct max patch.
>>
>
> it is quite hard to tell between 44 and 48 but if the 48 has ben
> made from the 96 you should hear the difference with good speakers
> and converters.
> with 192 it is another story; this often sounds worse than 96 (at e
> least with motu/rme/digi without big ben)
>

A couple of years ago, I saw a presentation from a guy who was doing
surround sound promotions sponsored by Tannoy. IIRC, he claimed
there was a study done showing that it was almost impossible to tell
the difference between material upsampled from 48kHZ to 96kHz and
material recorded at 96 kHz. Curious to hear about your results once
you get it working.

Higher frequency resolutions allow for better representations of both
the waveform shape (reducing the squaring out of high frequencies
near 22k) as well as the phase. Upsampling improves the
representation of the waveform which reduces distortion, but
presumably has less impact on phase? (there are still two possible
phase positions for a hypothetical 22kHz wave, but now they are more
spread out)

As someone mentioned before, though, it can make a difference in
processing with plugins.

Peter McCulloch

#115158
Nov 1, 2007 at 7:26am

Peter McCulloch schrieb:
> A couple of years ago, I saw a presentation from a guy who was doing
> surround sound promotions sponsored by Tannoy. IIRC, he claimed there
> was a study done showing that it was almost impossible to tell the
> difference between material upsampled from 48kHZ to 96kHz and material
> recorded at 96 kHz. Curious to hear about your results once you get it
> working.

Never trust statistics you didn’t fake yourself… If I can easily hear
the difference, the study was maybe targeted at untrained ipod degraded
ears… ;-)
Or they used high end converters which I would never be able to afford
in my whole lifetime…

Stefan


Stefan Tiedje————x——-
–_____———–|————–
–(_|_ —-|—–|—–()——-
– _|_)—-|—–()————–
———-()——–www.ccmix.com

#115159
Nov 3, 2007 at 9:33am

There’s a paper in JAES (Journal of the Acoustic Engineering Society)
September 2007 that might be of interest:

E. Brad Meyer and David R. Moran: “Audibility of a CD-Standard A/D/A
Loop Inserted into High-Resolution Audio Playback”.

Qouting the abstract:

“Conventional wisdom asserts that the wider bandwidth and dynamic range
of SACD and DVD-A make them of audibly higher quality than the CD
format. A carefully controlled double-blind test with many experienced
listeners showed no ability to hear any differences between formats.
High-resolution audio discs were still judged to be of superior quality
because sound engineers have more freedom to make them that way. There
is no evidence that perceived quality has anything to do with additional
resolution or bandwidth.”

The authors suggest that the reason why SACD and DVD-A recordings are
generally perceived to sound better is not the improved quality of the
digital format, but rather the difference in approach to the mastering
process. These recordings are assumed to be listened to in a hifi
setting, and the mastering is thorough, and using proper headroom. In
contrast most (popular) recordings are increasingly compressed to death
to fit in with the lofi listening situations resulting from modern
playback devices being used in noisy environments (cars, iPods, etc.).

Best,
Trond

Stefan Tiedje wrote:
> Peter McCulloch schrieb:
>> A couple of years ago, I saw a presentation from a guy who was doing
>> surround sound promotions sponsored by Tannoy. IIRC, he claimed there
>> was a study done showing that it was almost impossible to tell the
>> difference between material upsampled from 48kHZ to 96kHz and material
>> recorded at 96 kHz. Curious to hear about your results once you get
>> it working.
>
> Never trust statistics you didn’t fake yourself… If I can easily hear
> the difference, the study was maybe targeted at untrained ipod degraded
> ears… ;-)
> Or they used high end converters which I would never be able to afford
> in my whole lifetime…
>
> Stefan
>

#115160

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