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Human ear can perceive music from average power level from 0dB (-20dB in quiet places) to 120dB without feeling pain (for health reason, say we limit that to 100dB). That is 100dB range of volume setting. The music itself needs somewhat 16 bit representation, that adds another 100dB, so technically human ear can perceive a 200dB dynamic range.
16 bit is 96db. You cannot add that to the range of human hearing. You need to represent the range of human hearing within those bits....... Hearing damage occurs if you listen to sounds louder than 80db for long periods, 85db for shorter, and for over 90db you better start using hearing protection.
Everything you wanted to know about hearing damage but were afraid to ask:
https://www.osha.gov/dts/osta/otm/new_noise/
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there is still much room to improve. 16 bit is simply not enough -- that's why studios use 24 bit instead.
It's enough for playback. Not enough for mixing/recording.
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Human ear can perceive music from average power level from 0dB (-20dB in quiet places) to 120dB without feeling pain (for health reason, say we limit that to 100dB). That is 100dB range of volume setting. The music itself needs somewhat 16 bit representation, that adds another 100dB, so technically human ear can perceive a 200dB dynamic range.
16 bit is 96db. You cannot add that to the range of human hearing. You need to represent the range of human hearing within those bits....... Hearing damage occurs if you listen to sounds louder than 80db for long periods, 85db for shorter, and for over 90db you better start using hearing protection.
Everything you wanted to know about hearing damage but were afraid to ask:
https://www.osha.gov/dts/osta/otm/new_noise/
That doesn't somehow sound correct. It should be able to show a math formula expressing your explanation?
Strictly speaking, 16bit is 98db.
When I was talking volume range, I meant rms volume of sine wave. You need more bits to represent sine waveform, don't you? So the total dynamic range should be volume range AND snr of the waveform itself. -
From casually browsing their websites, both Zu Audio and Peachtree are "woo-woo" audiophool purveyors (I was going to use a stronger word.)
Zu Audio, for example claims for their "premium" Ethernet cable" "Music was more relaxed, information and instrument color was easier to hear but treble and attack was not emphasized. Still, presence was more vivid in a natural way, and bass was both clearer and had better weight." This is for a DIGITAL bitstream of ones and zeroes mind you. Their lack of technical understanding is more than counterbalanced by their vivid imagination.
Since we are talking about gear far removed from the mainstream, any statistics we gather from them are just as questionable as the underlying data. -
I didn't say you should be, I'm just saying you can probably build a respectable audio amp using those and retail it for under $100.A quick google search for "class ab 100w audio amplifier IC" came up with this in the first few results.
Well, somebody may be perfectly happy with that. I'm not, as simple as that.
My question to you is: How much would an audio amplifier you'd be happy with cost on the street?
You're an amplifier designer, I'm not. That's why I'm asking.
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From casually browsing their websites, both Zu Audio and Peachtree are "woo-woo" audiophool purveyors (I was going to use a stronger word.)
In other words, nothing much has changed in the HiFi reviewing business over the last few decades.Since we are talking about gear far removed from the mainstream, any statistics we gather from them are just as questionable as the underlying data.
cnet isn't mainstream?
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What does cnet have to do with anything? I was looking at the vendors websites directly.
Not to say that cnet has any particular credibility, either. -
there is still much room to improve. 16 bit is simply not enough -- that's why studios use 24 bit instead.
It's enough for playback. Not enough for mixing/recording.
DAWs these days work with 64-bit floating-point math.
-a -
What does cnet have to do with anything? I was looking at the vendors websites directly.
cnet sent you to those websites.
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16/44 releases today are dithered - with noise shaped dithers tuned to our perception of low amplitude noise
with reasonable psychoacoustic weighting you get more perceived dynamic range than engineering calculated flat bandwidth noise integration
with good dither and noise shaping you can actually hear a fractional lsb sine tone at ~3kHz more than 10 dB below the flat noise 16 bit calc
the dither + noise shaping is tuned to give best spot noise there, pushing the added dither noise to >10 kHz where our hearing sensitivity is less
CD res could be better than almost anything that was ever initially on classic analog magnetic tape
"Loudness War" dynamic range compression however is still a big problem in commercial music releases - entirely due to artist/producer/marketing choice, not technology limitations -
I didn't say you should be, I'm just saying you can probably build a respectable audio amp using those and retail it for under $100.A quick google search for "class ab 100w audio amplifier IC" came up with this in the first few results.
Well, somebody may be perfectly happy with that. I'm not, as simple as that.
My question to you is: How much would an audio amplifier you'd be happy with cost on the street?
You're an amplifier designer, I'm not. That's why I'm asking.
1) Not that chip, it is not suitable even for a half-decent amplifier.
2) The 60W/ch integrated amplifier in my home system had a retail price about £200 in 1998 and was build in China, however I've designed it here in the UK. Another design of mine (40W/ch) from about the same year was made in the UK and sold for £250 retail. I guess that today we need probably to double these prices.
Cheers
Alex
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I would not bother to talk to anyone who still believes analog medias are better than digital medias, even CD is better than most best analog medias, let along SACD or lossless master record.
Hmm, I need to check few things before talking or not talking to you
. IMHO, "CD quality" (16bit 44.1kHz) is crap compared even to a humble cassette at a standard speed. High resolution digital can be good it done properly (and that is not easy). Main problem for me is that there are very few good quality recordings available on high resolution formats. The introduction of the CD essentially killed the music recording and mastering skills, and the digital processing killed the recorded music quality 
.
Cheers
Alex -
Silicon Chip magazine published a headphone amp project September/October 2011 and you can see a bit of the article on their legacy (old) website here http://archive.siliconchip.com.au/cms/A_112574/article.html if you click on a diagram you can see a slideshow.
It was republished in EPE "Everyday Practical Electronics October 2014" (<<--search argument) and November 2014
If nothing else it will maybe give you some ideas on what you want to do.
If I'll call that project over-engineered it may be an understatement
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Cheers
Alex -
I have buildt a few of this amp http://nwavguy.blogspot.no/2011/07/o2-headphone-amp.html and they do sound neutral to my ears and my Audio Analyzer agrees.
Johan-Fredrik -
"CD quality" (16bit 44.1kHz) is crap compared even to a humble cassette at a standard speed.
I'd put some money on you not being able to distinguish between a tape playback and a good digital recording of that tape playback. -
Floating point is good, but ... double precision? That's overkill. No microphone, ADC or any other source is going to output that resolution. Still, whatever floats their boat. I guess it's free these days.there is still much room to improve. 16 bit is simply not enough
It's enough for playback. Not enough for mixing/recording.
DAWs these days work with 64-bit floating-point math.
Floating point master, 16 bit integer playback? Seems good to me.
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16/44 releases today are dithered - with noise shaped dithers tuned to our perception of low amplitude noise
Dithering is explained in that video I posted.Hmm, I need to check few things before talking or not talking to you
. IMHO, "CD quality" (16bit 44.1kHz) is crap compared even to a humble cassette at a standard speed.
It also mentions cassettes at about the 11 minute mark, right before he gets into dithering
(short version: They're only about 6 to 9 bits, but tape hiss is actually an analog version of dithering so they sound better than the number of bits would suggest).
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"CD quality" (16bit 44.1kHz) is crap compared even to a humble cassette at a standard speed.
I'd put some money on you not being able to distinguish between a tape playback and a good digital recording of that tape playback.
I am sure that you would require a DBT and a statistically good result to part with your money - and I won't agree to that because DBT in audio is a con, probably the biggest one, forget poor snake oil purveyors 
. If you want my subjective opinion, here it is: the chance to distinguish between the tape and it's digital copy depends on the source and the quality of the original tape recording as well as on the tape deck used - if we are talking about a standard speed cassette and a "CD quality" digital. It would be fairly easy for a high quality cassette recording and deck, however (if I want to cheat) it could be even easier for a poor and very noisy recording, because 16bit 44.1kHz are not capable of reproducing an analogue tape noise accurately . For a properly implemented double speed cassette you'll need at least 96kHz 24bit to be competitive. There is a 96kHz 24 bit sample of a choir performance on my web site that I've recorded on a cassette tape four years ago. If you reduce the resolution to 16bits and 48kHz, the quality of that recording noticeably suffers to my ear. I've started to work again (after a gap of 15 years) seriously with tape recorders about seven years ago when I've realised that I am not happy with the digital copies of my vinyl collection, even in a high (96kHz 24bits) resolution. I would prefer to go with digital because of the convenience, but I can not do much about my emotional reaction from music, so I keep it in an analogue domain as much as I can.
Cheers
Alex -
I'd put some money on you not being able to distinguish between a tape playback and a good digital recording of that tape playback.
I am sure that you would require a DBT and a statistically good result to part with your money - and I won't agree to that because DBT in audio is a con, probably the biggest one, forget poor snake oil purveyors . If you want my subjective opinion, here it is: the chance to distinguish between the tape and it's digital copy depends on the source and the quality of the original tape recording as well as on the tape deck used
We'll let you choose the cassette, etc., no problem.- if we are talking about a standard speed cassette and a "CD quality" digital. It would be fairly easy for a high quality cassette recording and deck, however (if I want to cheat) it could be even easier for a poor and very noisy recording, because 16bit 44.1kHz are not capable of reproducing an analogue tape noise accurately
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Analog tape noise can be reproduced perfectly, it's mathematically identical to digital quantization noise. Watch the video I just posted. The relevant part is around the 9-12 minute mark but the whole video is worth a watch.
The process of dithering is just about reshaping the digital "tape hiss" into something more desirable.
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We'll let you choose the cassette, etc., no problem.
Who are "we"
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1) I am not playing these guessing games, I have no time for that.
2) I am not forcing anybody to share my point of view, I'm just expressing it. Feel free to disagree.Analog tape noise can be reproduced perfectly, it's mathematically identical to digital quantization noise. Watch the video I just posted. The relevant part is around the 9-12 minute mark but the whole video is worth a watch.
The process of dithering is just about reshaping the digital "tape hiss" into something more desirable.
I have enough experience with all kinds of D-A and A-D processes for almost 50 years, thank you
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Cheers
Alex -
If you reduce the resolution to 16bits and 48kHz, the quality of that recording noticeably suffers to my ear.
It would be interesting to do a nice ~22 kHz minimum/linear phase high order low pass which rolls off over a couple kHz (brick wall filters are a bad idea) with 48 kHz decimation, round to 16 bit numbers and re-interpolate to 96 kHz with a high quality Sinc filter for a low resolution high resolution file.
I'm not good enough at DSP to do it without spending too much time though :/ -
2) I am not forcing anybody to share my point of view, I'm just expressing it. Feel free to disagree.
Sure, but I have doubts about what people are claiming, eg. that 96kHz, 24 bits isn't good enough.
(If it really isn't good enough then you need to take a look at your equipment and/or process, not the bitrate).
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for headphone amp projects they are more often found with accompanying audiophoolery
but projects that have been posted, worked over, have comments in forums can be useful learning tools:
Forums - Head-Fi.org Community is an enthusiast site - with a diy sub forum
www.Head-Case.org -the obligatory spin-off "opposition site" of those critcal of head-fi's fanboyism and reputed commercial slant
Headwize is inactive but archived at HeadWize Library - Projects [edit: may have to dig on archive.org]
you can still read/search the design/build threads for several popular diy projects
some diy project collections - now getting dated but useful links:
HeadWize: DIY Workshop > Current DIY Headphone Amp Offerings
and
DIY projects, kits - Head-Fi.org Community[/quote]
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A bit off-topic however I feel I have to say that:
To design a good sounding electronics is not a trivial task. There is much more to it than appears to an electronics engineer who has little to no experience in that very special area. Just as a simple example - yes, transducers are in theory the most non-linear parts of the audio chain. However for that very reason the electronics facing those devices on both ends is difficult to design right. Even a "perfect" amplifier has to work not with a dummy resistive load but with a very non-perfect speaker. So the real importance is how these two devices work together (and that is not an easy thing to get right, believe me, I've designed a number of amplifiers, including some very good ones, and some not so good). It is like in any specialised area of electronics, as soon as you start to dig deep you have to change your perspective and to learn some nuances.
As an amplifier designer, how much should a good audio amplifier with, say, 70-100W of power cost on the street?
A quick google search for "class ab 100w audio amplifier IC" came up with this in the first few results.
Low noise ... 0.01% THD ... 6 Euros in one-off quantity on Mouser.
I suspect the 'amplifier' problem has been studied, the design compromises have been made and it's now available in a monolithic packages from a dozen different manufacturers.
In a way, yes. Monolithic amplifiers can be very good and have excellent performance/price ratio. Discrete amplifier designs are still almost always crap that simply won't last (e.g. class AB output stages with Ube multipliers)
In commercial applications class D is extremely important, too, of course. Not just for low end applications, but also for high-end applications. Today you will often find that even ICs designed for low end markets incorporate not simply PWM modulators but actually quite good SDMs. Class D can surpass 0.005 % THD+N today (actually, ten years ago, but let's not nitpick on time lines).There is a fuzzy crossover point where true pursuit of high-fidelity reproduction becomes a completely subjective fetish. Certainly a more socially acceptable fetish than many others.
Well said!QuoteIf you reduce the resolution to 16bits and 48kHz, the quality of that recording noticeably suffers to my ear.
You should immediately visit a university clinic. Someone there is probably interested in doing some hearing tests with you.
As a matter of fact, increasing the Fs from 48 kHz to XYZ kHz does not add any information in the audio band (<20 kHz) at all, and never has.
Calling bullshit. To, like, all points you make. Your dynamic range calculation is just utterly wrong (at 30 dB SPL average you are unable to perceive a 100 dB dynamic range). Studios, just by the way, use 24 bits or 32 bit float not because 16 bits aren't enough, but because dozens of post-processing filters simply add a lot of noise. Using a high resolution format keeps that noise waaaayy below anything someone can hear.QuoteWhen I was talking volume range, I meant rms volume of sine wave. You need more bits to represent sine waveform, don't you? So the total dynamic range should be volume range AND snr of the waveform itself.
Uh, no, that's not how digital volume control in practical DACs/SDMs work. They won't reduce the effective dynamic range until quite some volume reduction, typically -50... -60 dB, because they directly fiddle around inside the modulator and don't simply scale the input PCM stream (which, yes, would proportionally reduce dynamic range. So in systems where no DAC volume control is used 24 bits makes sense - it still doesn't make sense to distribute 24 bit media.) -
engineers use 24 bit audio converters in all consumer devices
Please cite a convincing number of examples. I find that almost impossible to believe.
Certainly there are high-end audiophool devices that use 24-bit DAC, not because they actually have any effect on the performance, but simply for "bragging-rights". And because with modern, mass-market monolthic technology, a 24-bit DAC costs only a few pennies more than 16-bit. Furthermore, remember that outside SACD (which not 1 in 10000 people has ever seen) there is no 24-bit CONTENT to play. Not to mention that every year there are a vanishingly small segment of the customer base who even listen to uncompressed audio of ANY word-depth.Quotenewly emerged HiFi devices started moving to 32 bits
Please cite a convincing number of examples. I find that almost impossible to believe.Quotemany high end audio gears have analog volume control to preserve digital bits
Are there designs that operate the DAC at "full-scale" and then adjust the audio level in the analog domain? That has been a widespread practice from the low end to the high end practically since day-one. It is true that some designers have come to understand the downside of controlling reproduction DAC levels in the digital domain and are returning to analog controls.