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Some commonsense


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10 hours ago, Ajax said:

I have often wondered how humans can "hear" the difference between a 16/44.1 audio file, that has been recorded at say 24/96, and expertly mastered using dithering and noise shaping down to 16/44.1, and the original 24/96 file.

 

 

The obvious reason, which accounts for most of the difference, is that the playback chain behaves differently when fed source in one format compared to another - the type of distortion heard varies, even though the musical content, from the original recording, is identical. Implementation strengths and weaknesses usually mean one format will do better than another - most of the good stuff to listen to is "no better" than 16/44.1, so the answer is to optimise for that format.

 

"Fake" hi-res is actually superior to the "genuine" article, because the signal to noise ratio of the latter is very poor in the ultrasonic frequencies - graphs showing lots of activity at those inaudible frequencies are highly misleading, because if you actually look at the waveform, closely ... it's just noise. Pure noise. No matter what's happening musically, say, almost pure silence, or a grand crescendo, there's a contant ultrasonic hiss which never changes. Which IMO is the main suspect for why "real" hi-res sounds different - the electronic circuitry is reacting to the presence of this constant, inaudible noise; the distortion spectrum is different.

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37 minutes ago, Rexp said:

Most CD Players sound bad to me and I've never heard a good sounding Jazz CD, who is at fault? 

 

The people who design CDPs, and those provide the advice on what is required for achieving a good sounding system - jazz recordings are not a problem in themselves, but the typical instruments used by musicians playing this sort of music can very easily make for uncomfortable listening, if not reproduced well. I don't have a single jazz CD that can't be coaxed into providing a satisfying presentation, if I do enough to resolve SQ issues in the electronics chain - and this covers something like 100 years of recording this sort of music.

 

Get it right, and the impact of the music making takes your breath away - get it the tiniest bit wrong, and you'll hate the sound of it ...

 

Hardly anyone takes enough notice of the saying, "the devil's in the details" - but that's where the answers are ...

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1 hour ago, PeterSt said:

Notice that the square sound I talk about will be synth originated so it requires old fools like me to play that (kind of music) to begin with. Or Play AC/DC what you may be used to, like us with that guy living maybe 40 kilometers from here. But ZZ-Top is also fine for it. Or Rammstein. Or anything with a nice distortion guitar which, mind you, sounds beautiful when rendered as should (go figure about that because that won't go without amplification, so comparing with on-stage and the on-stage will lose ...). The distortion guitar is full with air and most will not have heard that (it requires "ultimate" speed of the whole playback system).

Now where were we ... Ah right, HiRes may not be needed.

 

Can only agree with that. Hendrix is quite sublime when replayed well - I have a copy of the Blues album here, and this is a fabulous ride of sound ... every twitch of the fingers, effects unit hissing, Marshall amp gurgling is laid bare; with a driving rhythm second to none.

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6 hours ago, audiobomber said:

CD's 96dB, seems like it should be enough, yet my experience is that, all else being equal, more bits sounds better. DSD resolution is equivalent to 20 bits in PCM, but DSD recording sometimes imposes a softness that I don't like.

 

My experiences run counter to just about everyone else's - over 3 decades ago I managed to optimise CD replay so that "everything made sense" in the replay - at the time everyone was jumping up and down, bellowing about how bad CDs were in conveying the 'musicality', etc of the performance - and I knew this was total nonsense. because of what I had heard from my own rig.

 

It was easy to understand why people formed that opinion; the standard was uniformly mediocre whenever I listened to a supposedly high end system, irrespective of cost - almost no-one was making the effort to further 'debug' their setups to the point where the sound snapped into shape.

 

6 hours ago, audiobomber said:

When first exploring computer audio, I downloaded a test that provided some familiar tunes, reproduced at 16/44, 24/48, 24/96 and 24/192. The biggest difference was between 16/44 and 24/48. Nevertheless, I believe the quality of the recording far outweighs the importance of 16 vs. 24 bits or sample rates beyond 44.1kHz.

 

I believe that oversampling or transcoding to a higher sample rate is beneficial, as it allows for a more gradual aliasing filter. I prefer to do this in exact multiples of the original signal to avoid interpolation. For 16/44, I use 176.4kHz (4X), or 5.6MHz DSD (128X). My Audiolab CD/DAC/preamp performs integer oversampling and upsamples up to 84.672MHz and 32 bits

 

The vastly most important factor is optimising the replay chain - doing that makes all the number games carry on evaporate completely, as being relevant.

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6 hours ago, tmtomh said:

Of course, if you reduce bit-depth, something is lost - but again, the result there simply is an increased noise floor from higher quantization error. This is the hardest part of digital sampling theory for our brains to grasp: "errors" or "missing/lost" stuff in digital sampling manifests simply as the noise floor, not as distortion or "gaps" or "stairsteps" in the waveform.

 

 

 

 

 

 

I've tried a few exercises of reducing 16 bit audio to 8 bits - oh, the horror!! - with nothing else changing, just making sure the best subjectively pleasant dither was applied - and it was just like listening to exactly the same track, in a somewhat noisy environment. I didn't feel that I had lost anything of the music; no wheels fell off in the process ... :).

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6 hours ago, Rexp said:

Agreed, I recently tried to downsample a 24/192 file to 16/44 and couldn't without ruining the SQ. 

 

Depends upon what you mean by "ruining the SQ" - if you merely played the resultant 16/44 using the normal method for playing that format you most likely are just showing that the replay chain behaves differently, depending upon format.

 

If I were doing such an experiment, I would use a high quality downsampling, such as that used in Audacity; and go the further steps of resampling the output 16/44 file back up 24/192 again. Using a simple comparison step, I could show that the only difference between the original and this new 24/192 was ultrasonic noise, plus a few tidbits of genuine music-related content above 20kHz, now and again. A final step could be to extract the ultrasonic content from the original file, carefully attenuate any non-noise content so that it never rises above the level of the surrounding noise, and add that to the 2nd 24/192 file, to create a 3rd 24/192 file.

 

So now we have the original 24/192 file; another with identical audio content below 20k, with no ultrasonic content; and a 3rd with something like almost pure ultrasonic noise, derived from the original. Now, in listening to these 3 using any sort of ABC procedure, can you pick them apart?

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14 hours ago, sandyk said:

Besides which,whether you like it or not, every format conversion like you are suggesting does result in some audible degradation, and is NOT a LOSSLESS process like many may wish to believe. Even doing the conversions  with different S/W such as Audacity or Sound Forge will result in small audible differences between the conversions.

This even applies to different S/W performing a simple conversion from 24/192 .aiff to 24/192 .wav files.:o

 

Alex, as mansr has been pointing out, format conversions to a precision level way beyond what can be considered audibly significant are quite easily achieved. The differences that are audible are all about the hardware being used for replay, and the precise manner of the playing of the tracks - tiny, usually ignored factors, or ones that are considered to be of minor importance are where the meat is, and vastly more relevant to the perceived SQ.

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38 minutes ago, sandyk said:

 

 Like many other members here, I prefer to use my ears , not look at instrumentation beforehand to tell me how something should sound.

 Neither do I accept that the players used with Audacity or Sound Forge  sound as revealing as JRiver 25 when playing from System  Memory.

 

Audacity works well enough to be able to pick differences that are audible - I've mentioned a number of times a simple technique that makes pinpointing audible variations quite straightforward, that I've used for years. Roughly align the the two tracks if necessary, select a promising area in the clips of a few seconds or less, and solo one track on repeat - a monotonous pattern of sound is created, like a mantra; as soon as you've tuned into that pattern, switch to the other track, using solo again. It's immediately obvious that the second pattern has very different qualities - if in fact there's an audible variation; going back and forth makes it easy to confirm  that a true difference exists.

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24 minutes ago, jabbr said:

 

Why? The issue of ultrasonics is not settled in my mind. No doubt there are fantastic CDs. But preservation of every bit of the recording preserves the possibilities of future as yet unknown technologies. That's common sense to me.

 

As someone who has heard how fantastic "very ordinary" CDs can sound when played on a rig with largely inaudible significant flaws, I see very little value in this. Vastly more is to be gained if understanding on how reasonable cost systems can be optimised to achieve the best reproduction from current releases was more widespread, IMO.

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An amusing side note ... I've compared the spectrums of YouTube audio retrieved at the best quality with an original track, and one can barely notice any variation, only starting to be significant at about 19kHz on. Comparisons of "CD quality" recording loops sometimes show more variation in the spectra than one what sees in those ... :P.

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8 minutes ago, Paul R said:

 

 Can not say I have done an exhaustive survey, but the half dozen or so recording studios I have personal knowledge of are all recording at at least 24/96k these days. Why not? There is no financial advantage to recording at a lower rate. That includes audio mixes for video. 

 

Some interesting comments here, https://www.soundonsound.com/sound-advice/q-should-i-use-high-sample-rates.

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2 hours ago, jabbr said:

With all due respect to those who insist that 16/44.1 captures all that we could possibly hear, none have you have presented to me an audio reproduction system which sounds entirely realistic, to me, and in my eternal hope that future audio reproduction systems will improve on the current state of affairs, common sense tells me to preserve every bit of a recording --  holding out the real probability that we will need new types of recording, yet nonetheless.

 

Said more technically: y'all are entirely forgetting nonlinear mechanisms.

 

The answers for achieving "an audio reproduction system which sounds entirely realistic" have always been available - but they are largely ignored because they're "not sexy enough" :). Not reeking of enough bling, and/or not having brilliant technical measurements are the usual suspects trotted out, as "essential requirements" - and are, yes, 100% BS ...

 

Overall system integrity is as dry, as exciting as your local suburban accountant - but unfortunately, ^_^, is at the heart - it's just tooo hard tracking down all those "weakest links", so it is almost never carried out to the necessary standard ...far more exciting to get hold of that supa dupa, turbo charged whacka tweeter - which is gonna solve all your problems ... :P.

 

Yes, it is nonlinear mechanisms - lots of them; tiny, tiny gremlins ...

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14 minutes ago, Paul R said:

 

Hint: more realistic audio is not going to happen without recording at a higher sampling rate. 

 

I agree it is not the only thing that needs to happen, and recording and playback are different operations. 

 

Closely guarded secret (:D) : highly realistic playback of the most unlikely material, going back over a hundred years of recording, is possible - if one does all the right things ... :)

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2 hours ago, Paul R said:

None of which will have zero value samples recorded in the digital file. You are talking about processing primarily for playback. In which case, as I said, the filters can be gentler and easier for high sample rate files. 

 

What the filters do to the waveform should be separated from the electrical effects of circuitry doing the realtime filtering, at the time of playback - these are very distinct factors.

 

An excellent example of this was an ambitious rig I heard, whose CD player had switchable filtering - "Which filter do you prefer, as I go through them ...?" - "Umm, none of them are doing it for me, theres a slight unpleasantness to the sound which is a constant for each one; is it possible to completely switch off this filtering function?" - "Huhh, unusual request ... okay" - "Ahhh, that's it! The "musicality" is in the room now ... "  ^_^

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1 hour ago, Paul R said:

But- I was taught that the set of  integers and their properties are pretty much at the root of all mathematical branches of study. Never really thought about it, just accepted that as an axiom. Of course, pretty much everything to do with integers is simply common sense. Sometimes incredibly obfuscated, but still...

 

.

 

Highly convoluted playing with maths may be necessary to understand, or distinguish what's happening between different examples of playback - but it's of zero use for actually evolving competent playback ...

 

So, rest easy on that front, ^_^.

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