The Myth of 'High Resolution' audio

[Audio Advent]

I think Stan has a problem with the overall volume in his glass at the pub gig compared to at home and the drinks cabinet.

[Audio Advent]

The idea of replacing all of the recordings I have in 16/44.1 with the same recording at a higher sampling rate is something I don't regard as 'easy' especially since 1) I don't do downloads and 2) it is often impossible to discover if the download is a genuine 24/96 recording or a 16/44.1 that has been upsampled.

Not sure why you'd feel the need to replace anything. For me, high res is all about future recordings which have actually been recorded at high res with top notch gear. Hopefully labels won't be shy in saying how it was recorded (you can probably look up the studio and see what gear they generally use and make a guess).

The music of the past in your collection can just remain as it is. I guess you could make an argument for pulling out some master tapes and digitising them at highres, but you're certainly not going to hear the original recording in high res, all you will get is simply the digitisation of a tape which is now decades old with all its top end loss, playing on a different machine with different setup and all sorts of imprinting (or whatever it's called.. ghosting?). Whether that's something one wants a copy of is up to the buyer - in some senses it adds it's own character like a decomposing, mouldy cheese. Might as well get the CD if it was made 20 years ago from a much fresher tape (but I'd favour the vinyl anyway probably).

[Audio Advent]

My reading of the Meridian guy is that he actually says we need 18.2bits to fulfil the potential of human hearing (in terms of volume range) and this is rounded up to 20bits to allow headroom for mixing and processing, he also suggests that a 58kHz sample rate is enough but argues that, out of convenience to provide simple convergence with currently used rates this could be up-rated to 88.2kHz (2x44.1) or 96 (2x48kHz)

In my mind, if you want to record something so that you definately end up with a result which fulfils the potential of human hearing in it's replay, then I wouldn't sail close to those limits, I'd go far beyond them to be sure. So for Meridian Man's assertions, I'd take it to 24 bits and 174 or 192 kHz. The only limiting factors to consider would be cost of the same quality converters at the higher rates (192 is pretty standard at the high end now, other than Dan Lavry's products) and data storage costs which are small now.

[Rothchild]

As observed, I think the jury is out on the sonic benefits vs potential disadvantages of 192. (although I'd agree that seeing as we have 24bit and it's way more than we need then stepping 'back' to 20bit is a moot point and just there for academic interest) - I do all my recording at 24bit and make use of the ample headroom (tend to peak between -18 to -12dBFS on individual tracks - also makes interfacing with analogue outboard easier).

One practical benefit to recordists is that, with a suitably powerful computer and well written soundcard driver etc 192 would enable you to have sub-millisecond recording latency through the box (with effects monitoring) which finally means that computer recording is at the point that tape was at the end of the 70s.

[mkrzych]

Hello,
I've been listening to high resolution files for quite some time and after all of these years I must admit that first if the quality of sound has been prime object and if master files are the same, the difference with redbook vs PCM 24/96 is subtle and not always is so simple judgement to pay more for high resolution.

However, if especially big hall orchestra or good studio jazz recordings are considered and are recorded and processed all the way down in 24/96 than you may have hear the difference. This also depends of your audio chain equipment like source and speakers/cans mostly.

Now, where to look for difference. In mu humble opinion first is spaciousness and things like particular instrument timbre and its decay - piano for instance. Second is the instrument placement in the stereo image and potentially feeling of wider soundstage. Those differences may vary as well for each particular recording.

Examples: Try Dvorak cello concertos by Steve Isserlis on Hyperion in CD and 24/96. Or recent Keith Jarred remasters, especially Belonging album. Comparing older CDs with that is nite and day truly! But in this case could be mostly influence of better tape transfer, of course mastering and potentially at the end of high resolution format.

Oh, forgot to mention you must have pretty quite listening room (try during the nite) without ANY major issues like hum, power noise etc. Mine is around 30-35dB of noise. If not the case, listen over decent dynamic headphones with for instance CMII.

[Audio Advent]

As observed, I think the jury is out on the sonic benefits vs potential disadvantages of 192. (although I'd agree that seeing as we have 24bit and it's way more than we need then stepping 'back' to 20bit is a moot point and just there for academic interest) - I do all my recording at 24bit and make use of the ample headroom (tend to peak between -18 to -12dBFS on individual tracks - also makes interfacing with analogue outboard easier).

One practical benefit to recordists is that, with a suitably powerful computer and well written soundcard driver etc 192 would enable you to have sub-millisecond recording latency through the box (with effects monitoring) which finally means that computer recording is at the point that tape was at the end of the 70s.

I'd agree that subjectively the jury is out but that is the same with so many "improvements" in the audio world - only takes one person to say they can't hear a difference and to say everyone else is experiencing confirmation bias and the case is adjurned indefinately. The theoretical arguments for and against often come down to particular practice and then belief - really the matter should be decided subjectively, the basis of empirical science if the theory under scrutiny is about something sounding subjectively better to a human.. so the jury remains out as before.

The tape comment I'm not sure about, not that it really matters as it's off topic I suppose... still, I'll ramble about it anyway

Surely that's comparing slightly different things? To actually compare like with like, wouldn't we be monitoring off the tape read heads? So the delay would be down to the distance between record and read heads and the speed of the tape. For there to be 1 millisecond of delay between record head and read head, the distance between them would have to be less than 0.4mm (0.381mm) apart at 15 inches per second (38.1cm/s). In reality they are a good few cms apart.. To compare like with like, you'd have to take the monitoring feed from before the A/D process, tap into the filtering stage before the A/D which is not what happens.

I think what you're thinking about is the use of plugins and monitoring through them otherwise wouldn't you monitor through an analogue desk if you needed to? You need to set the levels in the analogue domain before the A/D conversion so hopefully whatever is setting those levels has a monitor out on it. Same will happen if using any outboard digital effects like a Lexicon PCM90 reverb or if any digital pedals on the guitar, or Line 6 Pod type amp cab emulation is used. Line a few effects up at a time and things will probably experience quite a lot of latency. I guess the trick is for any performer's monitoring to be on a seperate set of channels with seperate analogue reverb for any musicians who feel they need it in their cans..

Even in an analogue system though, roughly each foot a performer is from the microphone introduces a 1 millisecond delay.. And with no system in place at all, a 20 foot distance between two live musicians and they hear each other with a 20 millisecond delay etc etc

[Rothchild]

Yeah, I'm wandering ot with the tape comment ;-) probably should have said analogue as it's not about tape per-se.

Low latency is useful because you can give the talent a decent desk mix back to their cans, with a big old school studio it was much easier to give a monitoring mix back that's close to what's going to tape (without necessarily giving them the tape monitor - which as you observe is delayed by the distance between the record and play head). In practice less that 7ms tends to be acceptable and less that 3ms is pretty good - if we can get the 'straight through' latency down below 1ms that leaves some flex for some processing in the monitor mix (which adds up as you've pointed out) whilst keeping the overall delay down around that magic 3-7ms total.

Interesting though isn't it that were arguing about the relative merits of 50microsecond of timing error (one cycle of 20kHz) and suggesting that we need 10microseconds of accuracy (96kHz) for 'real' hifi and yet we might be happy to expect a musician to play their instrument whilst being fedback the signal from that instrument 100x slower than that. (practically it does depend a bit on what they're playing (drummers tend to be a bit more sensitive to latency in their monitors than other, I'll hold off the bassist jokes ;-))

[Audio Advent]

well... how musicians play and their relative timings is a different thing to capturing a soundfield and all the phase relationships which give our brains 3 dimensional cues. Many musicans are happy listening to other musicians on basic cheap hifi and not bothered about the mp3 quality - the musician's relationships with each other and the notes played are generally captured perfectly by low end recordings. High res and all that, IMO, is all about sound and not music. Good sound of course helps the music to be better received but is not entirely necessary, else I wouldn't get enjoyment from the car stereo..

[Liffy99]

This is where we get into those marginal grey areas which I accept do exist. That's why I mentioned the possibility that frequencies higher than 22KHz, whilst being inaudible to humans, may have an impact on the reproduction of lower frequencies. As you say the additional bits (dynamic range) are not doing anything useful and any case only relate to dynamics and not the 'quality' of the sound reproduction in any case.

The point of my OP was that hi-rez audio is not in any way like hi-rez television. I think it is important that enthusiasts should be aware of that as it is pretty basic and relevant information.

Aliasing may have an effect at supersonic frequecies (the reflection of artifacts above half the nyquist limit back down the frequency spectrum. But anti-aliasing filters sort this out by introducing a gap between that threshold and aliasing components. I think this is why 44.1khz was chosen to cover a desired 20khz upper listening range (2 x 20 = 40, the minimum Nyquist sampling rate needed and the other 4.1 to create the gap).

[Liffy99]

Best summay I have seen . . .

https://youtu.be/nLEhfieoMq8