please explain 192 khz and 24bit.

[vrpatilisl]

hi
very new in this, so can anybody please explain to me the how exactly 192 khz sampling and 24bit depth affect sound quality also how we imporoved from audiocd to audio Dvd,SACD, Also from Ac3 to true hd.
Thanx

[LoRd_MuldeR]

"192 khz" is the sampling rate. It's the number of samples per second. For example Audio CD's use 44.1 kHz, DVD's uses 48 kHz. "HD" audio may use even higher sample rates.

"24bit" is the number of bits per sample, i.e. the number of possible sample values. The number of possible sample values is 2^n, where n is the bid-depth. So with 24 Bit, we can have 2^24 possible sample values.


The original analogous audios signals is continuous in both, time and value. This means the signal can change "at any time" and it may change to "any value" (in between the minimum and maximum).

When the signal is digitized, it will be sampled and quantized. Instead of a continuous signal, we then have a sequence of discrete sample values. And each of these sample values is coded with a fixed number of bits.


According to Nyquist's Theorem, the highest frequency that can be represented (without aliasing) is half the sample rate. So at a sample rate of 44.1 kHz, the highest frequency we can retain is 22.05 kHz.

As the human ear cannot hear frequencies above ~21 kHz - for older people the maximum is even lower! - 44.1 kHz should be sufficient in theory. If higher sample rates actually help sound quality, is up for debate ;)

Using a higher bit-depth mainly helps to avoid rounding errors when processing the audio, because after each processing step the audio needs to be rounded/quantized to the fixed scale again...


See also:
http://upload.wikimedia.org/wikipedia/commons/thumb/b/bf/Pcm.svg/1000px-Pcm.svg.png

[vrpatilisl]

thanx lord mulder for quick reply. now a days i see LP Record are again sold in market, they are analogue so why they gaining market. Any advantage over audio cd.

[LoRd_MuldeR]

now a days i see LP Record are again sold in market, they are analogue so why they gaining market. Any advantage over audio cd.

Yes, the DJ can scratch with an LP, which can not be done with a CD :p

Some "audiophiles" also claim that LP's have a more "warm" sound, compared to CD's. But that's question of personal taste, I think.

And of course nostalgia is a big reason why LP's still have a market in the era of MP3 and iTunes ;)

[Ghitulescu]

now a days i see LP Record are again sold in market, they are analogue so why they gaining market. Any advantage over audio cd.

There is no advantage over CD no matter what one says. To be able to reach the CD dynamics one must have a very very expensive player (in 4 digits US dollar), perfectly aligned, and special, heavy (audiophile grade) LPs. Otherwise their dynamics falls often under 60dB (sort of 12b, the CD has 16b) and their bandwidth is under 15-16kHz (a CD has 20, human hear stops around 17). The CD is also more precise (for stereo).
The only difference is in the odd harmonics, some like them more than those of a CD. They call it a "warm sound". One can obtain the same warmness by using a tube amplifier with variable drive (to set the exact % of distortions).

If digital is so bad, why do they, the self-called audiophiles, rip their vinyls in FLAC 24/192 :) ? Blasphemy I say.

BTW, what did google say when you asked it what a "sampling rate" is?

[amtm]

The only benefit of older vinyl records over a lot of the current crop of CDs is that a lot of vinyls weren't mastered with the hugely limited dynamic range that permeates the current loudness wars. But then again, that has nothing to do with the medium and more to do with the mixing. All things being equal mastering-wise, a properly mastered CD is going to beat the vinyl fidelity-wise despite what any "audiophile" is going to tell you. Besides, any theoretical advantage that an LP has, and it is entirely theoretical, is pretty much canceled out by the stylus damaging all those fine grooves the first time you play it and the distortion and destruction to the wave form introduced by your speakers.

[Midzuki]

The only benefit of older vinyl records over a lot of the current crop of CDs is that a lot of vinyls weren't mastered with the hugely limited dynamic range that permeates the current loudness wars.

Some people already complained about the CDDA re-masterization of the olde vinyls much before the rise of the loudness wars. Generally speaking, they said the "CD makers" attempted to "fix" :rolleyes: the non-existent "imperfections" :rolleyes: of the original master records.

[vrpatilisl]

Is dolby prologic II convert normal stereo sound to 5.1?
Thanx

[LoRd_MuldeR]

Is dolby prologic II convert normal stereo sound to 5.1?
Thanx

Dolby Prologic (II) is a technique to store a multi-channel audio signal in only 2 channels (Stereo).

With the suitable decoder, the multi-channel signal can be restored from the Prologic-encoded Stereo signal.

Of course this can only work, if the given Stereo signal was mastered/encoded with Prologic...

And of course Dolby Prologic can't deliver the same quality as 6 discrete channels, e.g. AC-3 or DTS.

[Ghitulescu]

Some people already complained about the CDDA re-masterization of the olde vinyls much before the rise of the loudness wars. Generally speaking, they said the "CD makers" attempted to "fix" :rolleyes: the non-existent "imperfections" :rolleyes: of the original master records.

And even before the parents of the same people complained about the "metallic" sound of the transistors vs. the warm sound of the tubes. :p

[amtm]

Some people already complained about the CDDA re-masterization of the olde vinyls much before the rise of the loudness wars. Generally speaking, they said the "CD makers" attempted to "fix" :rolleyes: the non-existent "imperfections" :rolleyes: of the original master records.

Oh I'm not going to disagree with you that many of the complaints against CDs for the last couple of decades are mostly audiophile bunk. But there genuinely have been botched digital transfers so as I said, a properly mastered vinyl will, obviously, win over a badly master CD. But then again that's no different than saying a well-mastered LD is better than a poorly mastered DVD. Vinyl, in theory, can produce better sound, blah blah blah, but due to many using rather average quality materials, most people having low quality speaker system and the plain damage from the stylus itself to the record pretty much prevents that from being a reality.

But hey, let's purchase $4000 dollar interconnects for our overpriced hifi so that we can hear the pops and clicks faithfully reproduced for that "warm sound". :rolleyes:

[bnshrdr]

There is no advantage over CD no matter what one says.

While I generally agree that most audiophiles exaggerate how much better vinyl is, I have to agree in one scenario. Sometimes on albums where there are very few releases, a studio could have really botched the mastering job on the CD and brickwalled it to hell. Sometimes, it may have been mastered much better on vinyl (I have found this to be true very rarely, but I have).

[nevcairiel]

Thats not the CDs fault, which is the whole point.

[pandy]

thanx lord mulder for quick reply. now a days i see LP Record are again sold in market, they are analogue so why they gaining market. Any advantage over audio cd.


which of coz is funny because CD and vinyl use same digital master source nowadays (ie whole studio is digital - at the end there is DAC that create "vinyl, analog, warm and fuzzy sound")

With proper processing typical CD is able to provide around 120dB dynamics and reduce (or almost completely remove) all digitalization (mostly quantization errors) problems.

hi
very new in this, so can anybody please explain to me the how exactly 192 khz sampling and 24bit depth affect sound quality ... SACD

This one is also quite funny - netto bit rate for 192ksps/24 bit/ stereo PCM audio is approx 9.216 Mbps, for SACD netto bitrate is approx 5.8Mbps however common misconception is that SACD is more analog or provide higher quality sound - this one is especially funny when you know that same master audio source (digital one) is used to produce all subformats ie CD, DVD, DVD-A, SACD and Vinyl.

[Ghitulescu]

which of coz is funny because CD and vinyl use same digital master source nowadays (ie whole studio is digital - at the end there is DAC that create "vinyl, analog, warm and fuzzy sound")
That is the mistake that is propagated low and high by most editors - in particular when they compare the both (with a tendency to one of them - strangely enough, they shifted to the other end 20 years ago, even if then both the ADCs and the DACs were under-performing by today's standards). The master is the same - yes, but before the cut, each tape is once more re-equalised and the dynamics adjusted to fit the medium. One cannot store the studio dynamics of 90dB onto a medium that can do at most 70 (normally less than 60 with regular record players).
With proper processing typical CD is able to provide around 120dB dynamics and reduce (or almost completely remove) all digitalization (mostly quantization errors) problems.
The maximum dynamic of a CD is 98dB (considering the sinusoid of 65536 values).

[MatLz]

The maximum dynamic of a CD is 98dB (considering the sinusoid of 65536 values).IIRC, should be 96dB for 16bit ( 6dB per bit )

[Ghitulescu]

IIRC, should be 96dB for 16bit ( 6dB per bit )

Indeed, but not always.

That occurs only when the MAX bzw. MIN values occur exactly at the MAX/MIN of the sinusoid. Try to imagine what happens when the peak falls between 2 MAX values - the sinusoid may go, depending on the frequency up to +6dB, on most gear causing clipping.

A better explanation, with figures, can be found googling for 0dBfs ....

[diogen]

now a days i see LP Record are again sold in market... Any advantage over audio cd.Here is one explanation I read a while ago:
LPs distort the sound track during playback much more than CDs.
But unlike with digital sound, those distortions are "pleasing" to the human ear...

The hissing is something that gets really annoying over time...:)

Diogen.

[pandy]

That is the mistake that is propagated low and high by most editors - in particular when they compare the both (with a tendency to one of them - strangely enough, they shifted to the other end 20 years ago, even if then both the ADCs and the DACs were under-performing by today's standards). The master is the same - yes, but before the cut, each tape is once more re-equalised and the dynamics adjusted to fit the medium. One cannot store the studio dynamics of 90dB onto a medium that can do at most 70 (normally less than 60 with regular record players).

Hmmm my English is not so good and I'm not sure that i understand You correctly... sorry for that...


The maximum dynamic of a CD is 98dB (considering the sinusoid of 65536 values).

yes and No - yes if human ear will be linear system, no when we exploit human hearing system characteristic then quite easy we can achieve 120 or even 130 dB (or even more).

http://www.e2v.com/e2v/assets/File/documents/broadband-data-converters/doc0869B.pdf

[Ghitulescu]

Hmmm my English is not so good and I'm not sure that i understand You correctly... sorry for that...
Sorry, my mistake...
That is the mistake that is propagated low and high by most editors - in particular when they compare the both (with a tendency to one of them - strangely enough, they shifted to the other end 20 years ago, even if then both the ADCs and the DACs were under-performing by today's standards). The master is the same - yes, but before the cut, each tape is once more re-equalised and the dynamics adjusted to fit the medium. One cannot store the studio dynamics of 90dB onto a medium that can do at most 70 (normally less than 60 with regular record players).
Once the main mix is ready, it will go to duplication. There each sound engineer will "rearrange" the "sound" and the dynamic to fit the medium.
For instance - a MC (cassette) has some -45dB noise floor, with Dolby some 5dB more. Silent passages will be flooded by noise, these must be "raised" -> compression. Also its bandwidth is low enough (up to 11000 Hz, yes, not 14 not 19 not 22kHz, all these marketing values are obtained at -20dB) so the sound engineer must re-equalise it again.
- an LP has a little bit more, but like for MCs, not all the record players manage to get over 60dB, each extra dB might very well double the price. Also the LP is non-linear (RIAA curve) and sometimes the bass is a bit attenuated and the trebles a bit raised, to compensate for their loss at [repetitive] playback.
- the CD has plenty of dynamics, is virtually noise-free - yet it has problems in the trebles. The sound engineer will still compress the sound, since most customers will rip it to MP3 to have it heard on portables - they need a higher sound pressure because of the ambient noise

The second part was directed to those "experts" at magazines, that hailed the CD over the LP. This, when most people remember the sub-mediocre quality of both ADCs and DACs of the time. Today, even the most Chinese player has better DACs than most CD-players of the early 80ies, and of course, the quality of the record players decreased (I mean those for public, not for audiophiles). But the LP beats the CD.

So,
1980: - worse DACs, unripe digital technique, superb HiFi record players -> CD beats the LP hands down
2010: - much better DACs, mature digital technique, worse record players -> the LP is better than the CD

[pandy]

Here is one explanation I read a while ago:
LPs distort the sound track during playback much more than CDs.
But unlike with digital sound, those distortions are "pleasing" to the human ear...

The hissing is something that gets really annoying over time...:)

Diogen.

You can record CD in a way that it will be not distinguishable form typical vinyl especially when additional distortions (transport and pickup) are taken into consideration - in past there was few models laser analog turntables - those one can offer quality better than CD however they are easily outperformed by DVD Audio.
From at least 15 yrs CD recordings are free from typical digitalization errors - it is a myth about digital sound "distortions".

[pandy]

Sorry, my mistake...

nope, mine - i see that i understand You correctly i was only confused by tapes and etc.


Once the main mix is ready, it will go to duplication. There each sound engineer will "rearrange" the "sound" and the dynamic to fit the medium.

this is more related to the "philosophy" of work (ie each sound studios can have own unique policy on this - those high end audiophile recording studios avoid such practice however generally some optimization and tweaking for distribution medium is performed.


For instance - a MC (cassette) has some -45dB noise floor, with Dolby some 5dB more. Silent passages will be flooded by noise, these must be "raised" -> compression. Also its bandwidth is low enough (up to 11000 Hz, yes, not 14 not 19 not 22kHz, all these marketing values are obtained at -20dB) so the sound engineer must re-equalise it again.


My personal experience shows that this is more close to 50 - 52dB, with Dolby B there is around 6dB gain, with C around 10, for Dolby S difference is very big and i must say 78 - 82dB is quite good. Bandwidth is also OK and with good shape tape mechanism 15kHz can be achieved without problems (especially when HX or similar dynamic bias current regulation is implemented properly.


- an LP has a little bit more, but like for MCs, not all the record players manage to get over 60dB, each extra dB might very well double the price. Also the LP is non-linear (RIAA curve) and sometimes the bass is a bit attenuated and the trebles a bit raised, to compensate for their loss at [repetitive] playback.


story is more complicated - one limitation is medium itself, second all those things related to reading vinyl.
Some of those limitations can be quite easily over-passed, some not (and thus very expensive laser turntable)


- the CD has plenty of dynamics, is virtually noise-free - yet it has problems in the trebles. The sound engineer will still compress the sound, since most customers will rip it to MP3 to have it heard on portables - they need a higher sound pressure because of the ambient noise


Not all recording studios use dynamic compression - some of them trying to prevent natural dynamic only with safe margin (carefully mastered recording is able to use for example -12dBFS average level and go up in peaks to 0dBFS without distortions with acceptable level of dither level and proper noiseshaping and thus offer more than 120dB dynamics for CD)


The second part was directed to those "experts" at magazines, that hailed the CD over the LP. This, when most people remember the sub-mediocre quality of both ADCs and DACs of the time. Today, even the most Chinese player has better DACs than most CD-players of the early 80ies, and of course, the quality of the record players decreased (I mean those for public, not for audiophiles). But the LP beats the CD.

So,
1980: - worse DACs, unripe digital technique, superb HiFi record players -> CD beats the LP hands down
2010: - much better DACs, mature digital technique, worse record players -> the LP is better than the CD

I disagree with opinion that in past especially DAC's offer worse quality than current (to be honest monolithic DAC are better than so called single bit) - for few early years issue was that many engineers was lack of knowledge about digital recording - lack of knowledge not due their ignorance but from general lack of knowledge - until dither and noiseshaping concept was created - digital recordings suffer a lot particularly from truncation/quantization errors and from jitter. Quality breakthrough in CD is tightly related to understanding those concepts.

Today Vinyl is fashionable, trendy - rediscovering roots of music etc however for sure this is not about quality (or perhaps not about objective quality).

[Ghitulescu]

We discuss the same things but with different perceptions: I know of audiophile initiatives (of JVC and MFSL to name the most known of them), I was talking about the bulk, because the bulk is important, they provide the money.
A normal ferro MC might reach -50dB, but we are not talking about TDK/Memorex or whatever HiFi product, we are discussing the bulk tape the replication studios use. It's not the best quality. However, I've seen also Chrome original MCs (equalised at 120us, not to pose problems at playback) which indeed will reach the values you quoted. Dolby B (and definitively C and S) are generally avoided, as any maladjustment cause a lot of damage in the sound.
And yes, the poor experience of the engineers with the digital technique caused more "moral damage" to the early CDs than the ADC/DACs and analog circuits.

I personally hate the surface noise of an LP, also the clicks and pops (I don't live in a chip manufacturing factory ;)) and the size. I can live with the CD's shortcomings.

PS: my personal belief is that the return of the LP is pushed by the studios since they fail to stop the digital ripping of the CDs. Going back to analog means that ripping an LP would yield far more inferior results to the casual user and at a higher expense and workmanship. I am not talking here about people having turntables of 500-1000€ (to reach the 65-70dB noise floor) and ADCs of a similar value. I am talking about Average Joe, that has a normal record player (he paid 200-300€ for, but it's only 20€ worth) and a regular soundcard. Or even worse, an USB-turntable that records only in MP3.
http://media.elektronik-star.de/images/L/10005704_title_usb_plattenspieler_ion_power_play_lp.jpg
BTW, have any of you seen any digitizing audio device that doesn't "rip" in lossy formats only? The only one I know of is the Ion LP2CD but its stored WAVs can only be accessed by burning a CD-R-Audio..

EDIT: I knew I had it somewhere.
http://oi41.tinypic.com/e11zep.jpg
This is from the service manual of an AIWA I've sold long time ago. It states the non-obvious info for many.
http://oi44.tinypic.com/30kdg9g.jpg
This is from a Pioneer
http://oi40.tinypic.com/ndlbpd.jpg
And finally from Tandberg Pro Series, which states another worrying issue.

[pandy]

We discuss the same things but with different perceptions: I know of audiophile initiatives (of JVC and MFSL to name the most known of them), I was talking about the bulk, because the bulk is important, they provide the money.
A normal ferro MC might reach -50dB, but we are not talking about TDK/Memorex or whatever HiFi product, we are discussing the bulk tape the replication studios use. It's not the best quality. However, I've seen also Chrome original MCs (equalised at 120us, not to pose problems at playback) which indeed will reach the values you quoted. Dolby B (and definitively C and S) are generally avoided, as any maladjustment cause a lot of damage in the sound.


i'm describing typical case - for very good tape decks 58 - 60 db without NR can be achieved - typical Fe tape, good quality should provide at least 50dB, this one for tape cassette decks - for studio they use 19 and 38 cmps vs 4.76cmps, also width of magnetic track is bigger thus higher SNR - also noise is analog not digital thus there is signal + noise and chance to hear this signal (only level comparable to noise). Cr tape provide around 54 - 58db without NR.
I have few recorded cassette tapes - some of them are Cr (for example for sure i have http://en.wikipedia.org/wiki/Nina_Hagen_%28album%29 cassette)

Personally im a quite happy owner of the Sony TC-K561S
Specifiaction is:
57db SNR without NR for Fe type, 30 - 17000Hz +-3dB quite good, Dolby S +24dB for 1kHz - 81dB SNR is also OK.


And yes, the poor experience of the engineers with the digital technique caused more "moral damage" to the early CDs than the ADC/DACs and analog circuits.


yep, first recordings trying to elimante noise at all cost - this was straght way to introduce quantization errors - and this create this frequently raised "metallic" sound.


I personally hate the surface noise of an LP, also the clicks and pops (I don't live in a chip manufacturing factory ;)) and the size. I can live with the CD's shortcomings.

PS: my personal belief is that the return of the LP is pushed by the studios since they fail to stop the digital ripping of the CDs. Going back to analog means that ripping an LP would yield far more inferior results to the casual user and at a higher expense and workmanship. I am not talking here about people having turntables of 500-1000€ (to reach the 65-70dB noise floor) and ADCs of a similar value. I am talking about Average Joe, that has a normal record player (he paid 200-300€ for, but it's only 20€ worth) and a regular soundcard. Or even worse, an USB-turntable that records only in MP3.
http://media.elektronik-star.de/images/L/10005704_title_usb_plattenspieler_ion_power_play_lp.jpg
BTW, have any of you seen any digitizing audio device that doesn't "rip" in lossy formats only? The only one I know of is the Ion LP2CD but its stored WAVs can only be accessed by burning a CD-R-Audio..


This one is quite funny decent MC pick-up + proper shape stylus cost around few thousand $ - also quite funny fact about vinyl - the more precise from recording perspective stylus tip shape is the more degraded vinyl plate when played (at some cases after 10 plays vinyl should be replaced by new one with low wear tip profiles it is around few hundred time... but this is about very expensive stylus - i doubt that many of those neophytes is able to distinguish between Shibata and for example Van der Hul)


EDIT: I knew I had it somewhere.
http://oi41.tinypic.com/e11zep.jpg
This is from the service manual of an AIWA I've sold long time ago. It states the non-obvious info for many.
http://oi44.tinypic.com/30kdg9g.jpg
This is from a Pioneer
http://oi40.tinypic.com/ndlbpd.jpg
And finally from Tandberg Pro Series, which states another worrying issue.

All those examples are probably without bias regulation and to prevent tape saturation relatively low recording level must be set.

http://www.docs.sony.com/release/TCK561S.PDF

For bias regulation HX (later Dolby HX Pro) was used however i have somewhere in my archive, Russian DIY project with very aggressive bias regulation - it use kind of PWM with AFAIR variable frequency, channel independent bias regulation and provide much better frequency response - without introducing incompatibility - bias regulation is used to optimize writing conditions and it is transparent from playing point of view.

[Ghitulescu]

Personally im a quite happy owner of the Sony TC-K561S
Specifiaction is:
57db SNR without NR for Fe type, 30 - 17000Hz +-3dB quite good, Dolby S +24dB for 1kHz - 81dB SNR is also OK.

All those examples are probably without bias regulation and to prevent tape saturation relatively low recording level must be set.

No, it's the standard (DIN 45500) that prescribes -20dB for several types of measurements.

See the specs of the super ultra highend X4 -> hier (http://www.produktinfo.conrad.com/datenblaetter/350000-374999/350833-da-01-de-X4TECH_TP1000_CASSETTENDECK.pdf)

[Groucho2004]

IIRC, should be 96dB for 16bit ( 6dB per bit )

Indeed, but not always.

That occurs only when the MAX bzw. MIN values occur exactly at the MAX/MIN of the sinusoid. Try to imagine what happens when the peak falls between 2 MAX values - the sinusoid may go, depending on the frequency up to +6dB, on most gear causing clipping.

A better explanation, with figures, can be found googling for 0dBfs ....

WTF? A 16 bit system (which the Audio CD is) has a dynamic range of 20 * log(65536) = 96.329598612473982468396446311838 dB

There is no "but not always".

[pandy]

No, it's the standard (DIN 45500) that prescribes -20dB for several types of measurements.

See the specs of the super ultra highend X4 -> hier (http://www.produktinfo.conrad.com/datenblaetter/350000-374999/350833-da-01-de-X4TECH_TP1000_CASSETTENDECK.pdf)

perhaps DIN45500 perhaps something else, nowadays usually also level -20dBFS is used to measure performance for AD/DA circuits (can be easily multiplied by 10 to describe performance for 0dBFS) , however tape decks use -20dB due tape saturation for high frequency signals (tape response curve is not linear - to improve linearity high frequency bias current is used) - low level simply allows to avoid saturation.


WTF? A 16 bit system (which the Audio CD is) has a dynamic range of 20 * log(65536) = 96.329598612473982468396446311838 dB

There is no "but not always".

This is bit tricky - 1 bit sigma delta should have then only 20*log(2)? or perhaps 1 bit system can have more than than 20*log(2)?

Very good paper http://www.tcelectronic.com/media/level_paper_aes109.pdf - highly recommended.

[ramicio]

I can't believe people still talk about jitter with digital audio. In reality, analog has much worse jitter (but not called jitter), thousands of times or even hundreds of thousands of times more than these nano- or picoseconds everyone worries about. Jitter is not a concern. Modern day DACs reclock the signal to have zero jitter before it hits the actual DAC. This is purely what sets new DACs apart from what you found in the 80s and 90s, and why things like USB DACs with added circuitry to fake deal with jitter, and they cost hundreds and up into thousands, make me laugh when a good sound card is all you need. Onboard still doesn't cut it in my book, though. Spending more than $200 to turn a digital signal into an analog one is pure insanity, especially when these expensive DACs use the same actual DAC ICs as what you will find in the budget ones. The electromechanical devices matter the most, and then the amps second.

I still don't buy that 16/44.1 is enough. A signal of music does not apply to that silly nyquist bullshit. It's a theorem. Notice the word theory in that? It works on paper, but somehow these people who hold science as their religion think it transitions perfectly in the real world. It doesn't. 16 bits is fine if you have loud music that is constantly peaking-out, but a nice quiet recording well below 0 dBfs that doesn't even have peaks close to 0 dBfs will benefit from 24 bits.

It all comes down to cheapness of wanting to store stuff. A person can easily store 20,000 songs of Redbook audio on a 500 GB drive. I have 660 GB of space taken, with almost 20,000 songs. I'd say around 75% is Redbook, 24% in better-than-Redbook, and 1% in MP3 for stuff I just can't find anymore.

[amtm]

But...but...5 nanoseconds of jitter is going to completely destroy the soundstage! At least that's what audiophile nonsense would have you believe.

[Groucho2004]

This is bit tricky - 1 bit sigma delta should have then only 20*log(2)? or perhaps 1 bit system can have more than than 20*log(2)?

What does that have to do with the actual content of an Audio CD?

I feel like I'm in the twilight zone.

[amtm]

I still don't buy that 16/44.1 is enough. A signal of music does not apply to that silly nyquist bullshit. It's a theorem. Notice the word theory in that? It works on paper, but somehow these people who hold science as their religion think it transitions perfectly in the real world. It doesn't. 16 bits is fine if you have loud music that is constantly peaking-out, but a nice quiet recording well below 0 dBfs that doesn't even have peaks close to 0 dBfs will benefit from 24 bits.

First of all neither "theorem" nor "theory" mean what you think it does. Contrary to the popular definition, in science "theory" doesn't mean "wild guess". Secondly, a theorem is a statement that has been proven using things such as previously accepted axioms or other accepted theorems, etc. So sorry, but your attempt to equate the Nyquist theorem, or theorems at large, as being the same as being some sort of scientific religion is nonsense. If you think that the Nyquist theorem is wrong you can supply your proof of that, right? Because there is plenty of both theoretical, again this does not mean someone's wild ass guess, and practical evidence to back it up.

Edit: To add, if the Nyquist-Shannon sampling theorem is wrong, then how did multiple people independently discover and prove it?

[ramicio]

I was going to buy a Bottlehead Crack amp, paid for it and all, but was just waiting for them to ship it for more than a month. So in the meantime I wandered their forums. The first issue I had was when this one guy said that gold tarnishes. I immediately got warned for my confrontational attitude. The second issue was when the OWNER said that he could hear a difference between uncompressed audio and losslessly-compressed audio. I argued with him. Got warned. At that point I demanded a refund, because the owner obviously wants a community of delusional folks who will spend money at the drop of a hat. The latest escapade was with another person who claimed that could hear a difference between files stored on an SSD and a hard drive, and even went to go on further to say he could hear a difference between files stored on different kinds of hard drives. Then he went on to say that SSDs in a Windows system is bad because he is under the assumption that a file system in Windows needs to be defragged, and since Windows 7 turns off defrag scheduling with an SSD, because it literally doesn't need it, ever, that Windows on an SSD is a bad thing. He then claimed to have been working with this stuff longer than I am old. He then went on to say that he was going to ignore me from then on. Well I got warned for some supposed confrontational attitude about trying to correct this fool about his misinformation that he knows and is spreading. I definitely don't recommend giving this guy any of your money.

[ramicio]

First of all neither "theorem" nor "theory" mean what you think it does. Contrary to the popular definition, in science "theory" doesn't mean "wild guess". Secondly, a theorem is a statement that has been proven using things such as previously accepted axioms or other accepted theorems, etc. So sorry, but your attempt to equate the Nyquist theorem, or theorems at large, as being the same as being some sort of scientific religion is nonsense. If you think that the Nyquist theorem is wrong you can supply your proof of that, right? Because there is plenty of both theoretical, again this does not mean someone's wild ass guess, and practical evidence to back it up.

Theorem does not equal proof, therefore it only applies on paper under certain conditions. Something cannot be proven by other things that are proven by other things that aren't proven, and you get an infinite line of things that aren't proof. Nyquist can NOT reproduce the EXACT shape of the original signal, it can only get close.

We also already KNOW that ultrasonic harmonics DO affect how we perceive audible sound, even all the way down to bass. Just because you can't hear a single tone of ultrasonic sounds does not mean they don't affect our reality and senses.

Theory does mean wild guess until something is proven. They are an educated wild guess, but still a wild guess.

[ramicio]

What's even just as laughable is that everything anyone quotes all comes from the field of information theory.

[amtm]

Theory does mean wild guess until something is proven. They are an educated wild guess, but still a wild guess.

Wrong. They are not "wild guesses". Theories are nothing more than the interpretation of the evidence gathered. It is not a "wild guess". This is just nonsense ravings by people who like to paint science as some sort of religion to knock it down.

What's even just as laughable is that everything anyone quotes all comes from the field of information theory.

Yes, a field that has tons of evidence and proofs to back it up that you have yet to even disprove even a tiny fraction of. Yet you attempt to brush it all aside with some silly argument over a dictionary definition of the word "theory".

[ramicio]

If it could be proven in the real world it would be called "information fact". Please. Math can get close to what happens in the real world, but never exact. I don't need to disprove a THEORY, a THEORY needs to prove itself.

[amtm]

Also, if information "theory" is so unreliable you obviously don't trust or use cryptography, right? You do realize that it relies heavily on information "theory" to work, right? You also do realize that seismic exploration also uses information "theory", right? A highly reliable application of it, too. One can go on and on and on about practical applications that rely heavily on information "theory" and yet despite being a "theory" it all pretty much works out as it's supposed to. Funny that, no?

[ramicio]

Nope, not funny. It's theory, in fact. Keep clinging to your Nyquist garbage thinking it applies to music or any kind of signal.

[amtm]

If it could be proven in the real world it would be called "information fact". Please. Math can get close to what happens in the real world, but never exact. I don't need to disprove a THEORY, a THEORY needs to prove itself.

If it could be disproven so easily you'd supply the evidence against it rather than blustering and broad brushing, right? You'd probably gain a noble prize for disproving information "theory".

[ramicio]

I don't need to disprove it. Anyone with common sense (anyone who doesn't hold science and math as their religion) knows that theory is not fact. If it's so magical, it should have no problem proving itself and having its name changed to "information fact".

[amtm]

Nope, not funny. It's theory, in fact.

Yet all the practical applications of information "theory" work just as the "theory" says it does. Sounds like it's a lot more solid then you'd have us believe.

Keep clinging to your Nyquist garbage thinking it applies to music or any kind of signal.

Again, not a single disproof of the theorem just more mud slinging. Also, yes, the original theorem was based on an idealized notion of having an infinitely sampled signal, but it doesn't mean it doesn't have plenty of practical backing to it.

[amtm]

I don't need to disprove it. Anyone with common sense (anyone who doesn't hold science and math as their religion) knows that theory is not fact.

More nonsensical mud slinging and broad brushing.

If it's so magical, it should have no problem proving itself and having its name changed to "information fact".

There is plenty of both mathematical proofs and practical evidence and practical applications of it to back up information theory. All you have is mudslinging and name-calling.

[ramicio]

Damn right. An ADC and DAC chain CAN NOT reproduce an analog signal to its original shape. Only an approximation, since it assumes everything it's being fed sinusoidal in nature.

[Ghitulescu]

WTF? A 16 bit system (which the Audio CD is) has a dynamic range of 20 * log(65536) = 96.329598612473982468396446311838 dB

There is no "but not always".

I cannot help but post a figure which is not mine (I'm on my internet PC that has nothing else)
http://www.solidstatelogic.com/music/X-ISM/images/Inter-Sample_Peaks_large.jpg

Imagine the 4 samples are MAX and MIN. See how the curve goes up to 1.1 (110%)? How much goes the reconstrued curve up depends on the frequency, or how close those 2 MAX samples are each to the other....

Your formula computes the dynamic of MAX-MIN, instead of computing the dynamic of the reconstrued curve ...

[amtm]

Damn right. An ADC and DAC chain CAN NOT reproduce an analog signal to its original shape. Only an approximation, since it assumes everything it's being fed sinusoidal in nature.

Not a single person in this thread has claimed you can get perfect reconstruction of the analog signal. :rolleyes: You seem to be arguing against something no one claims or has argued for. Everyone knows the practical limitations of the Nyquist sampling theorem. But once again, this disprove nothing nor is it any justification for the pathetic attempt at trying to claim that information theory is some field of wild guesses with no facts.

[ramicio]

Except that theory does not equal fact.

[amtm]

Except that theory does not equal fact.

*yawn* I've been suckered into this trolling for too long. So onto my ignore list you go and have fun with your lame dictionary arguments.

[ramicio]

Thanks for coming out. Have fun with your science and math. Have fun thinking theory applies perfectly to reality.

[Ghitulescu]

We are on this planet in our actual form for some 20.000 years. Yet we use mathematics to model the reality only for some 500 years. The cinema is some 130 years old. And see the 3D special effects in there. Leave us some time and we'll "fake" everything around us ;) and nobody will tell the difference :)

[ramicio]

If you don't believe that earlier civilizations weren't advanced and were just complete morons...

[MatLz]

WTF? A 16 bit system (which the Audio CD is) has a dynamic range of 20 * log(65536) = 96.329598612473982468396446311838 dB

There is no "but not always".
Yeah that was that.
For all calculations, the constant is log(2^20) which is approximately 6.0206.

Max dB == number of bits * log(2^20)

[diogen]

I've been suckered into this trolling for too long.Indeed.
But ain't it entertaining...:D

Diogen.

[Groucho2004]

I cannot help but post a figure which is not mine (I'm on my internet PC that has nothing else)
http://www.solidstatelogic.com/music/X-ISM/images/Inter-Sample_Peaks_large.jpg

Imagine the 4 samples are MAX and MIN. See how the curve goes up to 1.1 (110%)? How much goes the reconstrued curve up depends on the frequency, or how close those 2 MAX samples are each to the other....

Your formula computes the dynamic of MAX-MIN, instead of computing the dynamic of the reconstrued curve ...

The formula computes the theoretically maximum dynamic range for a given bit depth. The data on an Audio CD is stored with 16 bit, hence the dynamic range is ~96 dB.
You can "reconstrue" and interpolate as much as you want, it won't alter the dynamic range of the data stored on an Audio CD.

[IgorC]

WTF? A 16 bit system (which the Audio CD is) has a dynamic range of 20 * log(65536) = 96.329598612473982468396446311838 dB

There is no "but not always".
It's possible to obtain much better dynamic range for 16 bits (~110dB, if not more) with smart 24->16 conversion.

http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=40134&view=findpost&p=379082

http://wiki.hydrogenaudio.org/index.php?title=Noise_shaping
http://wiki.hydrogenaudio.org/index.php?title=Dither

[Ghitulescu]

I can't believe people still talk about jitter with digital audio. In reality, analog has much worse jitter (but not called jitter), thousands of times or even hundreds of thousands of times more than these nano- or picoseconds everyone worries about. Jitter is not a concern. Modern day DACs reclock the signal to have zero jitter before it hits the actual DAC. This is purely what sets new DACs apart from what you found in the 80s and 90s, and why things like USB DACs with added circuitry to fake deal with jitter, and they cost hundreds and up into thousands, make me laugh when a good sound card is all you need.
The jitter issue lies not in the cables or in the transport. Of course, provided the jitter is still within the allowed deviation.

It's in the DAC clock. Reclocking an old DAC (from an old CD-player) does improve more the sound than black CDs, "noise-free" power cables, audio cable suspenders and other wonder tricks sold for their weight in gold to stupid people with money that never heard the concept of placebo.

However, cheap [designed ancillary circuits for] DACs synch themselves to the incoming signal instead of buffering it and replaying it at a stable pace. For these, jitter in transmission is a problem.

[pandy]

I can't believe people still talk about jitter with digital audio.

...

I still don't buy that 16/44.1 is enough. A signal of music does not apply to that silly nyquist bullshit. It's a theorem. Notice the word theory in that?

OK, first jitter can be measured, there is very strong math behind jitter and jitter can be heard - theory and practice ALWAYS work - there is no practice without theory and theory without practice - they are complementary each other.

16/44.1 is acceptable from 99% listeners point of view, especially in real, natural listening environment - 16/88.2-96 is enough from human hearing system point of view.
With correct postproccessing (however before CD recording) 16/44.1 can match almost perfectly human hearing system.

What does that have to do with the actual content of an Audio CD?

I feel like I'm in the twilight zone.

Once again - dynamic digital audio is a bit tricky thing - without additional conditions dynamic of the 16 bit system can be higher than pure calculation with provided equation - human hearing system is not voltage meter - this very complex system and there is no simple and linear equation that describe this complex system.
i've asked about 1 bit system due of fact that 99% audio DAC use 1 or at "worst" case maximum 6bit conversion - so how this is possible that 1 bit DAC is able to provide 16 bit dynamics - taking literally You this can't work but it works quite well.


Edit: To add, if the Nyquist-Shannon sampling theorem is wrong, then how did multiple people independently discover and prove it?

There is very big area when non Nyquist conversion is used - ie sub Nyquist sampling, IF sampling, equivalent time sampling, random - stochastic sampling.



Nyquist can NOT reproduce the EXACT shape of the original signal, it can only get close.

...

We also already KNOW that ultrasonic harmonics DO affect how we perceive audible sound, even all the way down to bass.

Please do not be confused - Nyquist is OK but in PERFECT system (such system do not exist thus we need oversample) - i don't know why You introduce ultrasound and trying to use sub Nyquist sampling?

I don't need to disprove it. Anyone with common sense (anyone who doesn't hold science and math as their religion) knows that theory is not fact. If it's so magical, it should have no problem proving itself and having its name changed to "information fact".

Please... don't be so confused and please read about work of this guy http://en.wikipedia.org/wiki/Tertullian
He write something like this "Et mortuus est Dei Filius, prorsus credibile est, quia ineptum est. Et sepultus resurrexit; certum est, quia impossibile est" - This is exactly what are You telling...

The formula computes the theoretically maximum dynamic range for a given bit depth. The data on an Audio CD is stored with 16 bit, hence the dynamic range is ~96 dB.
You can "reconstrue" and interpolate as much as you want, it won't alter the dynamic range of the data stored on an Audio CD.

nope, this is not true, check this http://www.ti.com/product/pcm1774 - is Texas Instruments lie us when they tell that 1 bit DAC is able to provide 93dB dynamics? Or perhaps SACD is one big fraud - http://en.wikipedia.org/wiki/Direct_Stream_Digital literally says:
"The signal is stored as [...] a sequence of single bit values..."
how this is possible that SACD with DSD provide on only 1 bit 120dB dynamics? So where is error? Perhaps equation is wrong?

[Groucho2004]

nope, this is not true, check this http://www.ti.com/product/pcm1774 - is Texas Instruments lie us when they tell that 1 bit DAC is able to provide 93dB dynamics? Or perhaps SACD is one big fraud - http://en.wikipedia.org/wiki/Direct_Stream_Digital literally says:
"The signal is stored as [...] a sequence of single bit values..."
how this is possible that SACD with DSD provide on only 1 bit 120dB dynamics? So where is error? Perhaps equation is wrong?

I recommend you do some research on digital signal processing.

I repeat - What various DAC technologies do with the data after it is read from the medium has nothing to do with the actual data on the disk.

When the data is digitized for Audio CD, the dynamic range is "cut" to 96 dB because quantization depth is 16 bit. All the information that was between these steps is lost forever. The same applies to the time domain - all information between the sample points (~23 us @ 44.1 KHz) is lost.

Methods like integration, upsampling, etc. will increase the SNR and dynamic range but the resulting signal is still an approximation.

DSD is a completely different technology, it works with noise shaping quantization (which is basically pulse width modulation). Obviously, the formula for calculating the dynamic range of PCM signals does not apply.

[pandy]

I recommend you do some research on digital signal processing.

Appreciate Your advise however i prefer to wait for You.


I repeat - What various DAC technologies do with the data after it is read from the medium has nothing to do with the actual data on the disk.


I repeat - using equation provided by You and based only on Your knowledge i must say that DSD (SACD) can't work.


When the data is digitized for Audio CD, the dynamic range is "cut" to 96 dB because quantization depth is 16 bit. All the information that was between these steps is lost forever.


No - You are completely wrong with this claim.


The same applies to the time domain - all information between the sample points (~23 us @ 44.1 KHz) is lost.


Yes, but we talking about bit depth and digital audio system dynamics.


Methods like integration, upsampling, etc. will increase the SNR and dynamic range but the resulting signal is still an approximation.


Not sure what are You trying to say - can You be more precise?


DSD is a completely different technology,


No, why completely different technology - bit is exactly same like in CD audio system and there is only one bit so DSD (SACD) accordingly to You offer something around 6.02dB dynamic.


it works with noise shaping quantization (which is basically pulse width modulation).


You are wrong, PWM is something else than noise shaping, DSD doesn't use PWM at all.


Obviously, the formula for calculating the dynamic range of PCM signals does not apply.

suddenly bits are different?

Once again - how You explain that system with less than 16 bits depth is able to provide dynamics higher than 96dB?

How You can describe situation when i can provide to You signal sample - 16 bit, 44100 samples per second, LPCM, where i can store signal with level -130dBFS ie i can prove that on 16 bits i can have signal with 22 bit resolution and -140dBFS noisefloor?

[Groucho2004]

How You can describe situation when i can provide to You signal sample - 16 bit, 44100 samples per second, LPCM, where i can store signal with level -130dBFS ie i can prove that on 16 bits i can have signal with 22 bit resolution and -140dBFS noisefloor?

I can only assume that on your planet the laws of physics are different.

[pandy]

I can only assume that on your planet the laws of physics are different.

Or perhaps I've already done "some research on digital signal processing" and You still wait to discover and explore this very interesting world when theory and practice meets and work perfectly together.

[amtm]

There is very big area when non Nyquist conversion is used - ie sub Nyquist sampling, IF sampling, equivalent time sampling, random - stochastic sampling.

I know I said I wasn't going to respond anymore but this will be my last. Of course there is. The point is that if the Nyquist-Shannon sampling theorem were as bunk as ramicio is trying to make it out to seem there wouldn't have been multiple people to have independently have derived and proved it. That seems to be very wildly coincidental for multiple people to have derived and proved the same wrong thing. I'm not saying it can't happen but it's not very realistic. Also, if it were so bunk since it's only a theorem no one would ever bother with it. The fact that it is still relevant despite it's shortcomings, and there a number of them in practical situations, disproves the notion that it is worthless.

It's like trying to claim that Newton's Laws of Gravity are completely wrong just because they happen to break down at the quantum level. Both have their limitations but they are both still practically useful even when taking their shortcomings into account.

[ramicio]

You can't prove that the earth isn't flat and that the entire universe isn't just accelerating upwards at 9.8 m/s². Reality is based on observation and our understanding of that. Not absolute truths.

[Ghitulescu]

The formula computes the theoretically maximum dynamic range for a given bit depth. The data on an Audio CD is stored with 16 bit, hence the dynamic range is ~96 dB.
You can "reconstrue" and interpolate as much as you want, it won't alter the dynamic range of the data stored on an Audio CD.

We are discussing three items here.

The first one, which you are right about, is the dynamic provided by the numbers of 16b.

The second one is where I am right - it concerns the reconstruction of the original analog signal. -> http://www.tcelectronic.com/media/level_paper_aes109.pdf see figure 2.

The third one is the mapping of 16b into the electrical signal. The lower limit is the electronic noise. So, the higher the peak level, the higher the dynamic. So mapping 16b into a 18kV signal (electric trains supply) one can reach even 245dB. Conversely, mapping 24b into the regular consumer voltage (500mV, or -10dB), reduce its dynamic of 144dB to something less than 100dB.
The nicest part of this story is that there is no established method of mapping digital into analog.

[IanB]

16bit Linear PCM has ~96db dynamic range. It is by the very definition of the way the format is implemented.

Sure you can express more than 96db of dynamic range in a 16bit of data format. For example both the A-law algorithm (http://en.wikipedia.org/wiki/A-law_algorithm) and the Mu-law algorithm (http://en.wikipedia.org/wiki/%CE%9C-law_algorithm) provide schemes for encoding ~72db and ~78db respectively into just 8 bits. But there are tradeoffs for such schemes that use non-linear quantisation.

[IgorC]

16bit Linear PCM has ~96db dynamic range.
Noise shaping moves a quantization error to inaudible/less audible range (>20 kHz) during 24-> 16 bits conversion.

It means two LPCM files can have 96 db of dynamic range but sound totally different. Mathematically noise shaped LPCM still will have the same dynamic range than not processed LPCM but the first one will have better audio quality (as it had a higher dynamic range).

[LoRd_MuldeR]

I think this thread has gone off-topic long ago. So we will stop at this point.