digital - analog

<...At what point does sample rate equal the quality of analog?...>

What bandwidth of analog? 50-3500Hz? 0-20kHz? 0-50kHz?

Back in the the late 60's, long before digital audio, I studied human perception including psychoacoustics. It's the science of how people (and animals) hear. It didn't strike me as bull crap at the time, and the science had existed for a century prior to the 60's. The term was coined by Gustav Fechner in 1860.

Perfect and a GA from me.

I agree. The CD sample rate of 44.1 kHz is (more than) twice of the highest audible frequency which is 20KHz (the BW of the audio frequencies is 20Hz÷20KHz). Hence (due to physics) this sample rate is o.k. Of course, maybe there are some people (with "golden ears") who can hear slightly above the 20KHz. These people may sense an "intangible distortion-degradation" of the sound on the upper frequencies due to the sample rate... ...

I have a friend which is an audiophile and claims that he has "golden ears" able to listen to the intangible differences that anyone else can't hear. He got a Hi End "pure class-A" system with monoblock PAs and preamp (with elec. tubes of course) and super quality speakers (he spent a fortune for all these stuff)... That's fine so far... But the issue is that he spends, also, a lot of extra money buying many other "magical" things which are supposed that they make the sound even better (???) He knows very little about electronics so he is the perfect "victim" for those who sell such things. Sometimes I try to explain to him how things work and whether sth has a real effect or has no meaning at all. [I.e. he bought one meter of an exquisite (and reeealy expensive) power cable in order to power his system. He claimed that (using this cable) could hear an improvement on the sound (???) I told him that this 1 meter "super cable" is essentially useless if you consider the many meters of "bad" power cables running inside the walls all over his old building. After this argument he suddenly changed his opinion: he had the sense that (with this cable) the sound is worse (in contrast to his first opinion that the sound was better).] Sometimes another way to convist him is with a "blind test". With his eyes closed I connect or disconnect a part (which is supposed that makes the sound better) and he tries to decide if the sound becomes better or worse. This proves that this part has no real effect.

I believe that these people have a kind of obsession with the "perfect sound" issue. Most of it is just in their heads.

1. The human ear catches frequencies up to about 20 KHz. This determines sampling rate which is twice the bandwidth i.e. 40 Ksamples/sec. This says that any signal bandlimited to 20 KHz sampled at rate of 40 ksamples/sec can be fully recovered to original analog form.

2. How many bits you need to represent the sample. Usually 16 bits per sample i.e. 64000 digital levels is more than enough. Combining 1. and 2. the satisfactory bit rate is 40000*16 = 480 kb/sec.

CD digital rate is much larger due to error recovery. In CD they use error correcting Reed-Solomon code that require more redundant bits per information bit necessary for error recovery.

I do believe that most grownups have trouble with even hearing 16 KHz......!

That is true. However when you make system you should make it for everybody so that people who can hear more than 16 kHz are not excluded.

(I noticed a lot of either stupid replies or distasteful humor on this group. Please think before reply.)

A test we made at school many years ago showed that (in our physics class) less than 1 in 20 16 year olds could even hear 16KHz, one boy could hear to 18KHz.......

I am sure that most people composing music completely ignore these areas......

So why produce high frequency unnecessarily?It still costs power and equipment....

Low frequencies can be felt, if not actually heard, so producing them should still be useful....

The concept is that even if you can't hear those frequencies, they interact with lower frequencies... I don't know whether these interactions can be captured by a recording process that doesn't record these higher notes.

I think you are correct and that many of all ages don't know what listening is. They are only aware of audio and not intent upon it.

Andy,

Most of us OFs who worked in factories, foundries, mill etc have trouble hearing above 4kHz. Audiologists around here call it "mill hearing" and they are starting to see a resurgence of the same audiograms - except this time amongst under 30 folks who have never been in a mill. Tis the headsets, they think.

The sampling rate of twice the maximum frequency only applies for a steady state. An audio signal is fluctuating in amplitude, therefore a higher sampling rate is required. For mono the sampling frequency of 44KHz is probably adequate. For stereo i.e. two separate chanels, the sampling frequency should be doubled. I understand that systems using 96KHz do in fact give a better stereo image.

Tony Lee.

Your first statement shows that you do not understand sampling theorem. Sampling theorem fundamentally treats changing amplitude. Take any textbook or go to Wikipedia and learn it before posting the nonsense.

Your last statement is (not probably but certainly) correct simply because the highest bandwidth was taken to be 22 kHz. And for stereo you do need twice as much since stereo consists of two separate channels.

Actually, CD use about 20bits of precision. While this seems overdone on the linear range, it is not overdone in the logarithmic range. It takes a lot of dynamic range to reproduce the subtle sound of a flute playing beside a trumpet.

Overall, I agree that much of the sales speech for sound equipment is B.S. especially the cable business.

Actually audio CDs use 16 bits resolution.

Answer: ∞

yep when sampling rate is 1 to 1 but then you have analog again ;)

I had a fight with audiophile once, he tried to let me hear something he had re-recorded, and i could not hear in the music, what he said, so he became angry.

The audiophile believes that any time the recording has been touched by digital, the game is over 'cause information has been lost!

I know a couple that spent $10,000.00 on a friggin' record washer! And each time they play a record, they have to let it rest for at least 24 hours for the vinyl to come back into the correct formation.

I'm not even going to waste my time telling you how much they've spent on components!

well i could hear the difference on tube amplifier with those special loudspeakers that has a metal film with wires on each side of the fim with +signal on one side and - signal on the other (forgot the name), i took some of my ambient soundscape CD's like Lustmord And i heard suddenly a lot more sounds than i had on my big kenwood amplifier and my loudspekers.

it depends also a lot on the soundsystem. (and this guy built it all himself!)

The only issue I have with audiophiles is I think some lose sight of the fact that the purpose of listening is to hear the music. It's like being on vacation with a camera... If you get too much into the camera, you may miss the vacation.

For me, the big joy was when CDs came out - no crack and hiss and the fidelity was certainly an improvement. Cymbals and horns actually sounded like the real instruments. That much I noticed! But what really sold me was one day a friend and I were in a department store (Christmas time). Suddenly, we heard a violin solo from the audio department. It sounded pretty good, so I wondered over there to see what the name of the CD was. When I got there, I found a man who was actually playing a violin!!! That settled everything for me immediately. If a CD could sound like are real violin playing, then CDs were good enough for my ears!

I don't begrudge anyone their hobbies and how they want to spend their money. Good for them if they enjoy it. But I, personally now know my sonic limits, and would much rather listen to the music.

Oh and by the way, even I can tell the difference between really good and really poor speakers.

But the rappers and the samplers put the crack and the hiss back in

Yep, luckily they have cd players which can also simulate scratching

Well i have studio monitor loudspeakers and their frequency is larger than "normal" speakers. but the speakers this guy had (http://en.wikipedia.org/wiki/Electrostatic_loudspeaker) have a frequency range far beyond the human ear. and like you said, those unheard frequencies interact with each other and with the frequencies in the human range range, creating a whole new hearing experience.

i am just to lazy to built one myself(oh and you need to ad a sub woofer)

Yeah. I was interested in getting a pair of electro-static headphones. I called an upscale audio company and asked how much they were. He said they were $2,500.00. To which I replied, "What kind of a moron would pay $2,500.00 for a pair of headphones?!" There was a silence, then he said coldly, "I would." Oops!!!

time to make them yourself! done over 2years in the shed, in small increments

You will never satisfy an audiophile ..

'Cos they can hear stuff that they can't hear .

Del

Sampling is only half the problem.

If the reproducing circuit is not a quality design, my ears hear a difference.

I have tried the Monster cable/thin wire and there is a difference---mainly in power delivered, a minor change in frequency response, and distortion (very minor). However, I would hardly call it worth complaining about.

The original question was (I quote):

The quality of digital audio increases with sample rate. At what point does sample rate equal the quality of analog?

Or more practically.... at what point does sample rate satisfy the audiophile?

Will it ever? (satisfy the audiophile that is..)

So my answer (to the question), which is the correct answer, is:

One can fully reproduce the bandlimited signal if the sample rate is twice the bandwidth. This fully answers the question (no more story).

Your response relates to the quality of analog part, which is not a question.

So my answer (to the question), which is the correct answer, is:

This should probably read: "So my answer (to the question), which is partially correct yet seriously incomplete, is:"

That would better communicate the fact that you you've failed to answer the significant part of the question, namely: "at what point does sample rate satisfy the audiophile?

Will it ever? (satisfy the audiophile that is..) "

The sampling theory involved is well-known by many engineers, many laypeople, and (I suspect) by everyone who has responded to this thread. So in that sense, your answer added nothing to the thread: it simply parrots information that everyone already knows.

Why do audiophiles claim to hear things that others cannot? Is it placebo effect? Are they mentally unbalanced in the same way someone who hears "voices in his head" is? Are they insecure, and want to claim special abilities to shore up their egos?

Consider the "Kaptovator," a $1499 power cord for a hi-fi amplifier. Dave Clark claims to be able to hear the difference caused by using this cord versus another.

The Kaptovator vendor claims "No inductive or capacitive devices." That would seem physically impossible: every wire contributes to inductance, and any two wires separated by an insulator cause some amount of capacitance. They claim an "uninterrupted and speedy flow of power." Really? Are they claiming to alter the electron drift speed? AC power is notoriously noisy, variable, etc. This cord, by their own admission, does nothing to the power to condition it, has no shielding, and conducts all the garbage in your house lines directly to your amplifier. (With which any good power supply is perfectly able to contend.) You've got 100' of cheap wire leading to your wall outlet, which is fed by miles of power lines... and replacing the last 6' with ordinary wire is supposed to change something? Hmmm.

Dave Clark concludes:

• Are they overpriced? I have to say that after spending over four months with them, I am going to find it rather difficult to send them back. The Kaptovators are definitely worth the price of admission.
  

That audiophiles "hear" what others do not suggests that the interesting part of the question is the part you failed to answer. The part you did answer is simple high school physics.

Del's answer appears to me to be the most accurate and concise.

Read, I said READ, the original question and stick to the topic.

Here it is again:

"The quality of digital audio increases with sample rate. At what point does sample rate equal the quality of analog?"

So what is seriously incomplete in my answer?

Many people might know sampling theorem (you are not entitled to talk about others), however, it doesn't follow that you know it. Otherwise your discussion would be different. You are talking about hallucinations that do not belong to this tread (Electrical Engineering).

"Consider the "Kaptovator," a $1499 power cord for a hi-fi amplifier. Dave Clark claims to be able to hear the difference caused by using this cord versus another."

My bet is that Dave Clark would not be willing to bet $5,000 on a double blind test between the Kaptovator and a Walmart AC lamp cord.

Only fools would brag that they could tell the difference with a real double blind test.

I prefer digitally processed audio.

Analog is a process.

Digital audio occurs inside of software.

Digital audio can not be heard except as an analog process.

What is your question??

I don't debate how many ferries can dance on the head of a pin...I capture @ 320kbps / 44kHz...

at what point does sample rate satisfy the audiophile?

320kbs

That's a pretty cool fractal, bwire!!! Where did you get it?

Great answer!

I maybe on the wrong track here so forgive me if I go astray! Over many years, I've recorded bands and instruments using everything from 4 track analogue cassette to some large studio equipment costing some big bucks! I've met people who enjoy the natural compression of analogue which can add a warmth to the sound, and also people who like the crisp definition of digital! Both puritans to their preferred recording method! My take on this is put digital recording on duff speakers and it sounds duff, put analogue on duff speakers and it also sounds duff! Speakers are to the best of my knowledge all analogue, so however digital you go, you will always end at an analogue sound wave!

I don't think it's just a case of sample rate, more of how you prefer to here the music! Personally, I like it loud, clear tones (which I can get from both Digital and Analogue!) and full to the brim with lots of different frequencies!

So there you have it! No questions answered but it's a good topic! Just hope I don't get told off by 'you know who' for not answering in electronic tech talk! Hang on, I think I feel an electron flow moment coming on....No, it's gone, sorry!

Thanks all for the input. A great Tuesday morning read after a long weekend!

It appears that I need to update my policy. Topics to avoid at work: politics, religion and now audio quality.

Its not easy to answer, it most likly depend on musician's favourite.

Recently, the sample rate is up to 192KHz,/24bits. data stream is getting higher and higher.

in the past, 44.1, 48, 96khz rate/16bits and other rates are all existing in the HF, but 96khz is usually adpoted by people. 192khz chip is still expensive now.

However, to fit this situation there are two mainly coverter chip put into market in ordr that can change these rates mutually.

there is no limit for HF amature, but now most of instruments adopt 96khz/16b as thier standard. in he most time, advertisement effect is more strongly affacting teenis purchase.

Besides, there are many compress arithmetic to reduce date stream in the eqip;ment,

in fact, many audiophile cannt indentfy 44kz mp3 and 96mp4. in spite of all of them have heavy compress date rates. most of them feel very like original tape quality only but has some matel sound. when copmress ratio is up to 10.

Since no one has mentioned yet ...... Keep in mind that individual components in an audio chain must have appreciably wider bandwidth than 20-20kHz in order for the SYSTEM to have a bandwidth 20-20kHz. For example if you have a CD player, a preamp, an amplifier and speakers all with -3dB at 20 and 20kHz the SYSTEM will have a bandwidth of -12dB 20-20kHz. The -3dB points will be much narrower than 20-20kHz.

So it makes sense to get wide bandwidth components to preserve as much of that 22kHz source banwidth as possible.

yes . a cd or an audio equipment response frequency curve is tested from input (include preamplifier) to output, (uninclude speaker or earphone). -3 should be -3, if lower than this value, its would be marked actually value itself.

speaker would test individulal, it has lower features than elelctronic equipment. the late can even extent to 30khz ,40khz. or higher.

thats why good speaker is very expensive.

Overall system response is of course important, but CD players are not 3dB down at 20kHz, more like 0.5dB. Oversampling and decimation techniques while encoding and decoding have much more of a brick wall characteristic. No decent amplifier or preamp would be 3dB down at 20kHz so this isn't really an issue.

Specifying frequency response with respect to the -3dB points is a typical text book approach with no relevance to real world audio. (The German DIN standards once specified hi-fi quality as 20-20kHz ±3dB but this was irrelevant before they were published.)

In the broadcast world the entire signal chain has to meet the frequency response standards. (Standards vary from one organisation to another) Since the signal may pass though dozens of systems on the way, individual items need to be specified very closely and I would expect to see professional equipment with a response within ±0.5dB. Even so the cumulative error in an analogue system could be 5 to 10dB if uncorrected so there would have to be provision for equalising the signal. At least with digital the cumulative losses are eliminated as long as transcoding is avoided.

In the analogue world a typical music recording would pass through at least 5 analogue generations -

1. Multi-track session tape
2. Stereo mixdown
3. Mastering
4. Cutting the master disc
5. Final vinyl copy

Each of these systems had far greater frequency response errors than digital and required meticulous maintenance to keep the quality acceptable.

In a radio station audio from the disc would then be combined with other passing through the following stages:

1. Location recording
2. Edited version mixed with studio and pre-recorded material
3. Distribution copies
4. Playout through master control to transmitter via analogue landline or radio link
5. Final transmission
6. often a repeater station for fill in coverage using an off-air receiver or translator connected to a
7. repeater transmitter
8. Listeners radio

This list focuses on the recorders and radio links and landlines as they were the most difficult items to keep aligned accurately. We already have 14 critical steps not counting the various distribution amplifiers and switchers along the way. Digital replaces many of the critical steps making it easier to maintain the overall quality. It does illustrate the irrelevance of the 20-20kHz within ±3dB as a specification for equipment.

I agree that the audible frequency is limited as mentioned in other responses, however, what is not mentioned is the tone, the pitch of the signal, and the A/D, D/A converter ladders.

When processing an audio signal, the amplitude is restricted to specific amplitudes in both the A/D and D/A converters represented by a digital number. If the original analog signal is in-between these numbers (and it usually is), it will round up or down and distort the signal. Invariably the reproduced signal looks like a bunch of steps which is averaged with a filter. The quality of the signal is more affected by the voltage levels differences in the steps of the A/D, D/A converters then it is by the sample rates. Also the range or "number of possible steps" is a major factor.

Any distortion of the signal may be picked up as a tone variation. This also changes which harmonics are heard or not heard. Anytime you square off a signal you add harmonics.

Since some humans can detect variations in tone as little as 1/10th Hz, It results in a quality requirement 10X the the audible frequency. It also requires the digital word representing the waveform to be equal in bit representation of that frequency which means better A/D and D/A converters. Only when these conditions are met will digital be considered equivalent to analog (for audio listening purposes only.)

a/d d/a have been considered above.

you see 1 bit, 16bits is related to them. we call it quatitize precision. sample rate is different from this concept. sample rate should be according to Nyquist principle. in fact, you can use only samples in your audio equipent without a/d d/a converter.