44.1 vs 48 vs 88.2 vs 96

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peder
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Ah, Mea Culpa.
I missed the "not" in the quoted sentence :)

soybalm
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I think there is a difference between 88.2k and 44.1. It has to do with subharmonic distortion. When 2 different notes are played like a perfect 4rth. The harmony produces an additional lower note that is quite audible. The note produced is a 5th below the lower of the 2 original notes. I would think that this would create the need for headroom for high frequency harmonic content so the proper subharmonics can be produced which would better reflect real sound. These subharmonics could produce a whole other series of natural artifacts. Bob Ludwig insisted that mastering should be done at 24 bit, 96k. The music that I record is not that critical and I'm not that fussy so I record at 44k, 24 bit, but I bet a conductor or a real audiophile can hear the difference. I think I can hear more punch and solid imaging at higher rates but I am quite deaf to be honest from playing electric guitar every weekend. I noticed that some interfaces like to work at certain rates. Some gear will work at unsupported rates like my cheesy Creative xfi. It doesn't support 44.1 for recording but it does it anyways with sputters, dropouts where as my Apogee runs smooth as silk at any rate that I throw at it.
A western tuned 3rd produces the 4rth below the root so you end up with a chord. i think some cool stuff happens up there at 24k or so that effects 16k for example.
thanks

paul
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@soybalm: subharmonics are not an issue related to sample rate. as has been mentioned previously in this thread, the chief difference with going to higher sample rates is the nature of the brickwall filter that is used to prevent aliasing distortion (where higher frequencies get wrapped around to appear in the bottom of the frequency range). the higher the sample rate, the "better" the brickwall filter can be which means less aliasing which means less distortion of the original sound.

as for claims that people can hear the difference: there still (to my knowledge) are no double blind studies that show this, only a lot of noise from people who have never done a double blind study.

John E
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With a Nyquist frequency of just above 20KHz x 2, a 20KHz sine wave can theoretically be reproduced in its entirely from as little as two samples. Any other 20KHz waveform cannot (well, possibly a square wave) - but that's beside the point. At these high frequencies the human ear can only hear sine waves anyway. To be able to decipher a square wave or triangle wave etc at those frequencies requires that the human ear would need to be able to hear the relevant harmonics (in other words, the ear would need to be able to hear frequencies well in excess of 20KHz - which it can't). In audio, the Nyquist reconstruction of waveforms close to the cutoff frequency explicitly relies on the fact that the original signals can't possibly contain harmonic content which is capable of being heard.

But there's a catch.... when the sampling frequency is only just above 2 x the highest required frequency, practical reconstruction (i.e. decoding) of high frequency sine waves needs to rely on the fact that filters "ring" if you pulse them at frequencies close to their cutoff frequency. Thus, the quality of the reconstructed signal is very heavily dependent on the quality of the filter design. At a higher sampling frequency, filters can be used within the part of their spectrum that doesn't rely on ringing (but instead, can rely on having a larger number of samples from which to reconstruct the waveform). Increasing the sample frequency can thus reduce stress on the decoding filer which should theoretically improve high frequency linearity. It's quite possible that this might (might) be audible to a few people.

It's very unlikely though that 48KHz gives any better audible performance that 44.1KHz, given the same replay hardware but I can at least provide the answer to this point:-

From Seablade:

48kHz was used for syncing to film, I can't find a good article explaining exactly why and am not sure I could explain it very well from memory at the moment,

48KHz is mostly favoured because it offers improvements for film and video frame rates. 240KHz (48KHz x 5) divides to a very high degree of accuracy by 24, 25, 30 and 29.97.

Gmaxwell
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At least from the perspective of the converters it shouldn't in theory— but there is a practical situation where it does: A lot of consumer audio hardware (esp cheap sound cards) can only operate at at 48KHz, I presume due to cheapness and the fact that the two rates aren't related by a small integer so it's difficult to derive them from a single oscillator. To play 44.1k material these devices will resample, usually with a very low quality interpolation (see— previously mentioned cheapness). Running these devices at their native rate can greatly improve the output quality.

This isn't quite true. Although the final conclusion is roughly correct, these devices do not resample in hardware. Perhaps it makes no functional difference to users that its the device drivers that take care of it, not the hardware, but it does have the important consequence that the quality of the resampling is malleable. If you were using the right software, for example, it would never deliver material at 44.1kHz to a 48kHz device, but would have resampled it already, perhaps using a very high quality algorithm to do so. On the other hand, it is true that the quality of the sample rate conversion found in the basic device driver framework on Linux (ALSA) is not very good. But then again, its entirely possible to use different SRC if desired, and bypass the default stuff.

joegiampaoli
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A lot of consumer audio hardware (esp cheap sound cards) can only operate at at 48KHz, I presume due to cheapness and the fact that the two rates aren't related by a small integer so it's difficult to derive them from a single oscillator. To play 44.1k material these devices will resample, usually with a very low quality interpolation (see— previously mentioned cheapness). Running these devices at their native rate can greatly improve the output quality.

Yes, this has crossed my mind few times. I currently use an M-Audio Fast Track Pro and the specs are:

Digital Audio Interface Specifications > 48kHz sampling rate unless otherwise stated

http://www.m-audio.com/products/en_us/FastTrackPro.html

So I should get better response from it running it at 48 KHz which would be its native sample rate right? Not forcing it or stressing it to resample the audio during tracking, mixing.

EDIT: hmmmm, post I replied to has been deleted?

anahata
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There's another reason for using higher sampling rates for digital processing.

Any process that introduces non-linearities creates harmonics. Deliberate distortion generators are extreme cases, but more common examples are dynamic range processors: compressors, expanders, limiters. While the gain is being changed, sidebands are added to the signal, and a rapid gain change may add sideband material that is above Fs/2. In the digital domain (or strictly speaking for any sampled system), those products are aliased back into frequencies below Fs/2, and when Fs is 44.1kHz you only need harmonics above 24 kHz and the alias products have a good chance of baing audible.

The worst example is clipping, which if it happens in the digital domain can result in strong and very unpleasant sounding and harmonically unrelated alias products that wouldn't be there if the same clipping was done in the analogue domain.

Use of high sample rates moves these alias products to higher frequencies and would need much more severe distortion with higher frequency harmonic energy to produce audible aliases.

This is also the reason why analogue inputs should be clipped in the analogue domain before hitting the antialising filters - so if clipping does occur, the out-of-band harmonics are removed by the antialiasing filter. Not that there's any excuse for clipping a 24 bit input stage, of course :-)

Eq and reverb operating normally don't generate the harmonics that create this trouble; it's mostly dynamic range processors.

joegiampaoli
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And if I do want to track @ 44.1 KHz but this interface works internally @ 48 KHz then I should enable dithering in jackd, is that correct?

linuxdsp
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@joegiampaoli: Dithering is a method normally used to compensate for the inaccuracies caused when converting to a different resolution (bit depth).

joegiampaoli
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@linuxdsp:

Thx for your response.

I totally got messed up there for a sec. :) sorry 'bout that. Yes, I do understand what the dithering is for.

Still just wondering (for knowledge not that I would do a crazy thing like this), If the audio interface would work at another depth than the one I want to track to, would enabling the dithering in JACK dither that audio in real time for the recording? And with good quality? If not then what precisely is the dithering inside JACK for?

Thanks.

linuxdsp
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Generally (as I'm sure you know) you would always want to track at the highest resolution (bit depth) you can, and I can think of almost no benefits to dithering audio while recording (except for possibly saving storage space, but in these days of huge disk capacity and system memory that's almost never an issue anymore - for pro audio anyway). As to what jack is capable of in that respect, I think its a question for Paul or the other jack developers (I'd have to look at the code anyway to see what dithering algorithm it uses, and I have no practical experience of ever using jack to dither audio at all - although it might be interesting to try out the options with baudline or similar audio tools and see what the effect is...)

paul
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@joegiampoli: there is no reason to use dithering in your scenario. as linuxdsp noted, its only use is when reducing the bit depth (e.g. from 24 to 16 bits). it exists inside JACK so that people working with a 16 bit interface (there are still a lot of them) can optionally turn it on for the final conversion of audio into 16 bit form as it leaves the JACK ALSA backend. it has no role to play within JACK clients, where all audio is and remains 32 bit floating point (though i suppose something that (foolishly) converted to and from 64 bit floating point internally might want to use some form of dithering as well). Clients that can export to a disk file (like Ardour) can also use dithering if asked to write a file with <24 bits of resolution, but they do not do this via JACK.

joegiampaoli
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Ok, thank you so much. I just wanted to know why there would be a dithering option in JACK.

Well anyway, I'm doing some TESTING now to see what results I get tracking @48KHz (Fast track Pro native internal sample rate) obviously in 24 bit mode, mixing at same values, then exporting (upsampling) up to 96 KHz 24bit for mastering and then dither it down to 44.1KHz 16 bit (a la Bob Katz).

I'll let you guys know if the results are really worth it. I've just seen too many posts on other places where they swear by this method, so I gotta give it a go to see for myself.

Cheers!

skiller
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hey hello
sorry about my wild english
this is not the truth ....only my experience

well this thing is very complex ( mathematically ) but in my experience , it depends of the kind of stuff you have... so if you have an
audio-interface that supports 44.1, 48 ,88.2 ,96 and 192 sample rate you must need to have a pair of speakers(monitors) that supports those sample rate ....but the thing about is :

between 44.1 and 48 there´s not difference really (what really matters is :if you are working at 24 bit or 16 bit.."".24 bit is much better but when you finish a song you need to render or export to 16 bit, many people use dither to avoid "loss" but some times it is not needed you have to listen and decide what is better)

...... 44.1 sample rate is for any kind of music-recording audio(standard) and 48 Sample rate is supported by video-editing-recording(cameras diginal-analog or video interfaces and soft. ) for this two options you only need a common pair of monitors ..... i mean analog(with rca inputs or input 1/2) it will sound good at 46 and 48 sample rate so although you increase your "interface audio sample rate" to 88.2, 96, and 192 you wont hear difference .......(IF you have speakers such i mentioned before(analog) you will increase the work on your processor (computer) and nothing will become better).........

well when you use 88.2 ,,96 or 192 sample rate you before have to know this:

44.1 *2 = 88.2 so this is for audio(music)
48 * 2 = 96 ............................(videos,films etc)
96 * 2 = 192.............................( digital optical audio environment) (between 88.2 an 96 Khz there´s not difference really ,, it depends of if audio is for music or video project )

so if you have an -interface audio- with "digital /optical out" and also you have speakers with "digital/optical input" and both support 88.2, 96 and 192 sample rate you really will hear the difference between "44.1, 48 sample rate playback" and 88.2, 96, 192 sample rate playback" (they have to be connected via tosh link optical cable""""" if you connect them via rca input or 1/2 input you wont hear the difference im talking) but it doesn't matter really because ''not everyone has/ digital/optical audio-system/ .....thus only
who has /optical/digital audio - system /will be able to hear the details of the sound at 88.2 or higher.

but the advantage of working at 88.2, 96, 196 sample rate for example during recording,, you catch some details that you didn't catch at 44.1 or 48 sample rate and when playback each sound has more detail.

the problem is that when you Finnish a track you can´t export it to 196 or 96 sample at 24 bit because there would be a lot of "loss" audio due to not everyone has optical audio system as i wrote before... normally we all export tracks to 48(video), 44.1(audio) sample rate and 16/24 bit so a kind of rule to avoid loss audio is/are:

for audio you work/record at 88.2Khz and 24 bit then you export tracks to 44.1 Khz and 16/24 bit(only few optical monitors supports 88.2Khz)

for audio-video you work/record at 96Khz and 24 bit then you export tracks to 48Khz and 16 bit/24 bit

about 192Khz and 24 bit can be used at sound installations or digital theaters for HD films with HiFi optical Audio(very expensive technology )) rare¡
also if want to record birds (or other animals) 192 khz is the best sample rate to do that ,,,96Khz as well. also if you want to record some inaudible/low.hi noises.textures or freqs to make experimental music works ok,,,,96Khz as well.

also
for video you can work/record at 192Khz and 24 bit then you export tracks to 96Khz and 24 bit and then to 48Khz and 16/24 bit (not common//only me)

(But If You Really Want To Avoid Any Kind Of " Loss Audio"............ Record And Playback Like This ....Music 44.1Khz,24/16 Bit And Video-Audio 48 Khz, 24/16 Bit...... Better 24 Bit For Recording And 16 Bit For Exporting

well its all i can say about .... bye bye