audiofools...maybe not so much

[AndyC_772]

Guys, do your research!

Point 1: Consider a simple SPDIF signal to DAC. The DAC timing has to run in sync with the input. Not just at the same nominal rate, but in sync. If you think about it, there is no other way.

One word: FIFO. Your argument would only make sense if there were no such thing as dual-port RAM.

Provided that, on average, one sample is clocked out for each one which is clocked in, then a simple FIFO can be used to *completely* decouple the cycle-to-cycle timing of the samples coming from the SPDIF interface, from the cycle-to-cycle timing of those same samples going out to the DAC. All it does is introduce a very small pure delay, which may only need to be a few samples' worth.

You still need a clock to drive the DAC which runs at the same nominal rate as the incoming data, but it only has to match the long-term average rate. It can be as simple as a VCXO governed by a control loop that looks at the amount of data in the FIFO at any given time.

How do I know this technique works? Several years ago I designed a DAC, from scratch, which works in exactly this way. The DAC chip itself is connected directly to the output of a VCXO, and the VCXO is controlled by a chunk of logic in an FPGA which wakes up every few seconds, makes a tiny adjustment (if needed) to the VCXO control voltage to keep the FIFO about half full, then goes back to sleep again.

If you can spot a path by which jitter in the SPDIF signal could possibly affect the DAC clock in this topology, I'd really love to know what you think it is.

[NiHaoMike]

The newer digital receivers also have adjustable delay to compensate for lag on some TVs and a "game mode" that simply sets the buffering to minimum and turns off processing that adds significant lag.

[Sigmoid]

Niquest limit

Nyquist.

[Sigmoid]

Consider: Each stage must give out samples at the exact rate they are coming in. Otherwise you have to start inventing samples (local clock is fast) which would be awful. Or if the local clock is slow you'll lose lip sync, music plays a while after you hit stop on the player and no matter how big your buffer, eventually you run out of room in the buffer and have to start throwing out data. That is not good either. You have to run each stage in sync with the incoming signal.

[removed]
I see AndyC_772 has already said pretty much the same I wanted to. FIFO, transmission control, burst read/transit. You know, these things have been around for a while now.

In reality, CD players have a small buffer and the timing comes from a local oscillator. (Which in mediocre products, is polluted by having a motor in the same box and not taking adequate protection for the purity of the oscillator.)

Welcome to the real world, Neo.

That is a very common misunderstanding. We are so used to data transmission, that not many realize that digital audio relies on analog timing information carried along it (on bit edges).

I'll look into the timing aspects of SPDIF, I'm curious. Still, we weren't talking about timing. We were talking about "misaligned sensors" and light not hitting dead center. That, timing information or not, sounds absolutely ludicrous.

[JuKu]

If you can spot a path by which jitter in the SPDIF signal could possibly affect the DAC clock in this topology, I'd really love to know what you think it is.

Yes, I believe the scheme would work and you could make money with it. The only possible jitter is the inherent noise of the vcxo (insignificant with good execution) and the timing of the contol loop (fraction of a Hertz, again insignificant when done right). I believe the product would be rather expensive, though. Sounds very high-end from all sides. I'm not aware of any comercial product doing that (price, likely?). Are you?

(Edit: typo correction, iPad keyboard...)

[JuKu]

I'll look into the timing aspects of SPDIF, I'm curious. Still, we weren't talking about timing. We were talking about "misaligned sensors" and light not hitting dead center. That, timing information or not, sounds absolutely ludicrous.

That could affect the signal strenght on the sensor which affects bandwidth. I couldn't think any other explanation to the measurable jitter differences in receiver parts of same type and same manufacturer. If you have another idea, I'd like to hear it.

(Yeah, if you have a jitter sensitive system, input 1 might sound better (or worse) than input 2. )

But we are beating a dead horse. This issue was understood and solutions found at least 15 years ago.

[JuKu]

Jitter is boring, let's change the subject to the audibility of absolute phase: 

...but if they [speakers] are both using the same polarity, the difference is really only on attack and only on the first cycle.  its something very few people can truly detect with reliability.

Note that most waveforms are asymmetrical. I've heard demo signals where the absolute phase was changed, and I heard a change in the color of the sound. Maybe I heard it because I was supposed to, but the demo was set up to show that for some type of signals absolute phase matters. I've heard that female voice and trumpet are the natural sounds where the difference could be audible.

Maybe some people can learn sounds so well that they can hear absolute phase. Probably, a lot of experimenting is required as well to learn the difference. In real life there are so many other things affecting sound color (speakers and room the biggest) that I don't think I have any chance (outside an artifically set up demo).

[hamster_nz]

... There is very good reading about jitter at http://nwavguy.blogspot.fi/2011/02/jitter-does-it-matter.html. Do look at least the second image that shows how the encoding and limited bandwidth on the cable generates jitter. In the bottom are very real life examples: A motherboard (truly awful), a couple of popular cheap external USB DACs (better, but still audible) and a $1600 DAC (something for the designer to be proud of). Btw: I'm not the NwAvGuy, and I have no affiliation to any of those products.

Interesting graphs on the link - do I read it right that the jTest is a 'torture test' based around a 11.55kHz -3db test tone, and a spectrum analyser picks up noise with individual frequencies at around -100db or less depending on quality of the DAC?

But you are right, that with a better DAC when you are belting out 500W of power (+57dbm) the most energy in a jitter induced 'false' frequency on the truely aweful DAC is 0.1 microwatt (?40 dBm)? to 0.01 microwatt (-50dbm).

[JuKu]

... There is very good reading about jitter at http://nwavguy.blogspot.fi/2011/02/jitter-does-it-matter.html. Do look at least the second image that shows how the encoding and limited bandwidth on the cable generates jitter. In the bottom are very real life examples: A motherboard (truly awful), a couple of popular cheap external USB DACs (better, but still audible) and a $1600 DAC (something for the designer to be proud of). Btw: I'm not the NwAvGuy, and I have no affiliation to any of those products.

Interesting graphs on the link - do I read it right that the jTest is a 'torture test' based around a 11.55kHz -3db test tone, and a spectrum analyser picks up noise with individual frequencies at around -100db or less depending on quality of the DAC?

But you are right, that with a better DAC when you are belting out 500W of power (+57dbm) the most energy in a jitter induced 'false' frequency on the truely aweful DAC is 0.1 microwatt (?40 dBm)? to 0.01 microwatt (-50dbm).

You can't really read signal to noise ratio from a graph. NwAvGuy did not publish the number. I've seen my share of graphs with numbers attached. Just an educated guess, but I don't think -80dB is far as S/(N+D) for the motherboard graph. That is somewhere between a phone and well done CD.

[lewis]

now, I refuse to believe that usb cables would matter at all, ever.  from usb to the dac is always a packet exchange, its not a one-way pipe like spdif is.  the timing is derived from the pc (and not the dac, if UAC1 style) but still, a whole packet of data has to be received by the dac before its burst out and so, it does have local buffering.  timing is controlled by the pc but the per-bit timing, during the exchange, is not relevant.

USB audio tends to use the isochronous transfer mode - the packets get chucked out of the computer to the DAC with limited latency, but with NO retires on lost/corrupted packets. It is essentially a one-way pipe. There is a CRC but no way for the endpoint to request a replacement packet. The USB host also prioritises the Interrupt and control transfer modes, so if the bus is busy any isochronous transfers may be delayed. The reliability of the connection therefore becomes an issue, and any induced noise interfering with the data link could corrupt packets with no way for the endpoint to do anything about it.

You don't get corrupted data when you copy a Word document to a flash drive because file transfers use the Bulk transfer mode. In this mode there is no guarantee of latency (it can be as slow as you like), but there is full packet checking and retires on error. This is NOT the case with isochronous where "small errors are not catastrophic and can be tolerated".

I've never seen this explanation mentioned elsewhere, but this may explain the alleged audibility of some USB cables. Or it's just total bollocks.

 

[NiHaoMike]

USB audio tends to use the isochronous transfer mode - the packets get chucked out of the computer to the DAC with limited latency, but with NO retires on lost/corrupted packets. It is essentially a one-way pipe. There is a CRC but no way for the endpoint to request a replacement packet. The USB host also prioritises the Interrupt and control transfer modes, so if the bus is busy any isochronous transfers may be delayed. The reliability of the connection therefore becomes an issue, and any induced noise interfering with the data link could corrupt packets with no way for the endpoint to do anything about it.

You don't get corrupted data when you copy a Word document to a flash drive because file transfers use the Bulk transfer mode. In this mode there is no guarantee of latency (it can be as slow as you like), but there is full packet checking and retires on error. This is NOT the case with isochronous where "small errors are not catastrophic and can be tolerated".

I've never seen this explanation mentioned elsewhere, but this may explain the alleged audibility of some USB cables. Or it's just total bollocks.

A good solution is to put an indicator on the front of the DAC or converter that lights up if it detects a CRC error.

[John Coloccia]

And yes, these were two brand spanking new reverb tanks for a Fender guitar amplifier. Same specs, same manufacturer. Identical in every conceivable way. What got me going with him was that he came in with a new tank that he didn't like the sound of. Said it was wrong somehow and wanted me to scrounge up a NOS tank.  I was like come on you've got to be kidding me?! So I did an informal test with two accutronics tanks and every single time... BINGO, he knew it! By conventional wisdom it should have been insignificant, especially with a reverb tank that's used as nothing more than a mechanical delay line. Since then I've never dismissed anything off hand.

So obviously, they weren't identical.  Reverb tanks are electro mechanical contraptions.  I would EXPECT there to be a good amount of variance in it.

re: different caps
Did you carefully measure the actual value of the cap?  The tolerances on many caps are very wide.  Also, different cap construction have different frequency responses.  When all of the variables are controlled for and a real experiment is done, the differences always disappear.

Part of what I do is build guitar effects pedals.  None of them sound exactly the same.  There's always a little variability here and there, and that's with high quality, close tolerance parts.  Every now and then I build one that just doesn't sound right, and the board goes in the trash.  If I built these things with Radio Shack parts, they'd be all over the place and 50% would end up in the trash.

[Sigmoid]

You don't get corrupted data when you copy a Word document to a flash drive because file transfers use the Bulk transfer mode. In this mode there is no guarantee of latency (it can be as slow as you like), but there is full packet checking and retires on error. This is NOT the case with isochronous where "small errors are not catastrophic and can be tolerated".

I've never seen this explanation mentioned elsewhere, but this may explain the alleged audibility of some USB cables. Or it's just total bollocks.

Bit errors do not "degrade quality". They translate into blips and screeches.

Digital does not "degrade" like analog. It's either 100%, or it's 100% broken.

[JuKu]

Regular spdif audio can be affected by cables. As said many times, the precise timing rides on bit edges, and limited bandwidth inject noise to PLL. The cable could in some cases have an effect (most obviously through the quality of shielding in coax).

Isochronous USB needs a PLL too. I can see how an uniformed audiophile can convince himself that the quality of USB cable matters as well. I have harder time figuring out how cable could affect the PLL in USB.

[linux-works]

Bit errors do not "degrade quality". They translate into blips and screeches.

Digital does not "degrade" like analog. It's either 100%, or it's 100% broken.

this isn't a digital bit error, per se, but it is a digital audio error: ever hear a DAT machine mistrack?  back when I was involved with DAT some of us would use non-standard tape lengths (60meters or 2 hours of standard play was most common and in-spec) but 3 hour 90meter or even 120meter tapes were also found and meant for data DDS drives (which can adjust tension to make up for thinner tapes).  on audio systems, none that were made back then were able to auto tension and so every so often, you'd hear a mistrack and it would sound like a buzzsaw.  very annoying and ruining of the audio ;(

the cure was to keep the system aligned, cleaned and never use thin tapes.  it was not cheap to keep the machines always CLA'd (to use a photo term) and so many of us had to deal with the odd buzzsaw here and there.

I never saw errors over the spdif lines, no matter how badly I abused cables.  I built many spdif switches and buffers and media converters (toslink and coax) and used the worst possible cabling - but never heard any bit errors.  you have to really go out of your way to create data corruption in spdif, at the bit level.

at the timing level, sure, things are easy to get wrong, but the format (with reclockers at the rx side) is very forgiving.

[lewis]

Isochronous USB needs a PLL too. I can see how an uniformed audiophile can convince himself that the quality of USB cable matters as well. I have harder time figuring out how cable could affect the PLL in USB.

The PLL is nothing to do with it. If the packets get dropped, and with no way to recover them, the audio is affected.

[AndyC_772]

Yes, I believe the scheme would work and you could make money with it.

It does. I've been listening to it every day for about the last five years.

I believe the product would be rather expensive, though. Sounds very high-end from all sides. I'm not aware of any comercial product doing that (price, likely?). Are you?

The total cost is about £3 for the VCXO and maybe £10 for the FPGA, so the added cost of simply getting the topology right is negligible. Other than designers having a mental block about how a product "should work", I can't think of a single good reason not to design a DAC this way.

USB has been mentioned too, and my DAC does also have a USB input. It uses an off-the-shelf TI USB DAC chip which is designed to go into cheap sound cards - but as well as analogue, the chip also has an SPDIF output. I don't doubt that the jitter performance of the SPDIF stream is pretty poor, but of course, that doesn't matter in the slightest in this case. It all goes into the FPGA, through a FIFO, and emerges under control of the VCXO.

I did look quite seriously into making the design available commercially, but there's a huge amount of work involved in bringing a product to market over and above a clever schematic and a working prototype. Since I already have a job, and I can't see myself ever getting rich off yet another piece of niche hi-fi gear, I decided against it.

Instead, I got an upgrade to the sound of my hi-fi, and the satisfaction of knowing that what I'm hearing is coming from a design which is right in an academic sense. And, of course, that fact alone makes it sound better!

[JuKu]

The total cost is about £3 for the VCXO and maybe £10 for the FPGA, so the added cost of simply getting the topology right is negligible. Other than designers having a mental block about how a product "should work", I can't think of a single good reason not to design a DAC this way.

Welcome to real world: You just at least quadrupled a typical outboard DAC parts costs, and are on different planet for a receiver. For a typical $1000 street price receiver, the digital audio in section budget is <$1. Add some for the micro and bigger PCB, convert to $ and we might be at 24$. There are 23000 (yeah, every penny counts) reasons NOT to design run-of-the-mill consumer stuff that way. (This is why I don't want to do those; vast majority of my audio work has been high and even higher end.)

USB has been mentioned too, and my DAC does also have a USB input.

Does it do 96kHz/24bits?

I did look quite seriously into making the design available commercially, but there's a huge amount of work involved in bringing a product to market over and above a clever schematic and a working prototype. Since I already have a job, and I can't see myself ever getting rich off yet another piece of niche hi-fi gear, I decided against it.

PM me if you would want to talk about licensing it.

[auxie22]

Quite a number of years ago I use to have hypersensitivity disorder and struggled to be in a crowded room like a mall / shopping centre and most audio equipment to me sounded horribly "muddy" (what I know now is the "muddy" sound was actually Fcu roll off between 18-20k and so it seemed "muddy" to me at the time) or seemed strange or distorted because I could hear the nyquist frequency on CD's. I got my hearing tested and could hear between 20Hz-25kHz. I could hear the difference between most audio systems due to the difference in the extreme top end. Not long after that I became a Live Sound Engineer and started going deaf even though I used earplugs, the constant 90+ dB SPL's and not to mention age slowly brought the upper end of my hearing down to 16kHz and due to this I have had a much better time in large crowded areas and even with sleeping. Sometimes I think the "Golden Ears" aren't always a blessing. I know I struggled with it and am happier "without them". I can actually enjoy music on most audio systems now unless my analytical objective Sound Engineer side comes out .

[powerhouse]

Your title and post got my attention (also found the above post very interesting). I must admit that I'm not familiar with psychoacoustics, but I did and do listen to music using what some would consider audiophile equipment, and have listened to a variety of mid to high-end stuff. Oh, I used to be able to hear 20kHz (or even higher?), now it's at 16kHz or so. I can still make out lots of noises, high pitched sounds, etc. that most people including youngsters don't hear or notice.

To put it simple, for me the difference between enjoyable audio equipment to let's say less enjoyable hifi stuff is in the extent to which it engages me. If I get tired after listening to familiar music over  some time it means something in the equipment chain is not up to the task. If the music keeps engaging me then the audio equipment is worth closer examination. Selecting the right auditioning music also helps. Tomshardware gave an excellent example on how NOT to do hifi auditioning - see http://www.tomshardware.com/reviews/high-end-pc-audio,3733.html. According to their editor spending more than $2 in a DAC is a waste of money  . If you look at their method there is - in my opinion - no way they can tell apart the (near) true-to-life reproduction of music versus the average rendering of digitized music. (The reasons why I think so are: 1. TH uses mostly studio multi-track recordings that are impossible to relate to any real concert experience; 2. The auditioning periods are much too short, you can't make out much if any difference when listening to a 3 minute track on DAC1, then on DAC2, etc.; 3. They were trying to identify differences in what they hear, whereas I would be looking for what's missing; 4. It hasn't been mentioned, nor is there any reason to assume, that TH chose any music recorded in a concert hall that the auditioners were familiar with. When I audition audio equipment I bring with me at least one or two recordings where I am familiar with the recording venue and specific acoustic properties; 5. TH used stereo headphones which, as good as they are, can't really reproduce a natural sound stage.)

One reason why not-so-good audio equipment reproduces music in a way that exhausts me is the effort I'm forced to make to locate instruments/players on stage.

I can't tell if this has to do with the ASMR you mentioned, but I certainly can provide plenty of examples where even a "small" change in the audio setup (for example a different interconnect) can make a noticeable difference to me, even though it may be on the level of how I feel or how I enjoy the music, and not on any identifiable difference in the sound. In many cases I would need to listen to music for a prolonged time to become aware of it, and I doubt if the difference I hear (or better "feel") could be easily identified by test equipment. Let's put it this way, there are few if any specs that audio equipment producers publish that have any bearing on the quality of the music reproduction - even the price tag doesn't give an indication (though good stuff tends to cost more).

All my audio equipment and cables have been selected by me on the basis of purely subjective criteria: the level of pleasure it gives me when listening to music. In short, yes, I think there is a lot more to music and music reproduction than what many want to make us believe. ASMR could perhaps help to explain this.

[powerhouse]

Earlier in this blog a comment of not hearing in 5.1 was made,  the external ear gives to sounds arriving from different directions a different spectra,  the ear does a simple fft,  this is easily tested,  block one ear,  blindfold and have a friend click his fingers  in different positions,  typical accuracy is better than 5 degrees with ONE EAR,  While a lot of audiophile stuff is rubbish,  some of the really complex analysis has not been done. (All silicon PA at home here!)

I didn't say that isn't true. What I said is that good stereo can replicate everything 5.1 can. I've heard some pretty damn amazing stereo...

The reason 5.1 was invented was cinema. Stereo only works if you control the position of the speakers relative to the listener. It's essentially a one-person affair, and doesn't hold up in a cinema, with people sitting all over the place.

+1

[powerhouse]

You can see all there is to see from most windows, just like you can hear all there is to be heard with most hifi. But it takes an incredibly good and clean window to make you believe there is no glass.

Exactly !!!

I've read through most of the posts now and it always amazes me on how many people think that they fully understand the world and the human being and on how we hear and perceive music. Because I don't.

The only thing I know is what I like. And I like to hear a classic concert so that when I close my eyes I can hear the solo violin a little left centre, near the front of the stage, and it doesn't move or become fuzzy with the change of pitch or frequency or string played. And so on for each instrument in the orchestra, or singer. But this is only one of many qualities I expect to find in my audio gear. Measurebators will be hard pressed to put the qualities I'm looking for into numbers. And those companies who try to put audio quality into numbers can go look elsewhere for customers who are impressed by those - I'm definitely not!

[edavid]

I've read through most of the posts now and it always amazes me on how many people think that they fully understand the world and the human being and on how we hear and perceive music.

Straw man argument... you don't have to "fully understand the world etc." to know when an audiofool is spouting bullshit!

[DrGeoff]

The recordings are not made using perfect microphones.
They are not digitised using perfect ADC units.
The consoles that they are mixed on are not perfect. If analogue bus summing is used on an external console than it goes through another DAC/ADC step.
Then the mastering stage will require further passing of the signal through various analogue boxes (another DAC/ADC step). Finally you have your 16-bit dithered digital stereo file ready for CD impression.

So why would anyone think that buying excessively over engineered equipment make any real noticable difference? You could just buy an Apogee or similar DAC (as is used in the studios) to get the same output.

[linux-works]

most recordings are crap.  its so true!

there are very few that truly deserve 'special handling'.  its for those that the audiophiles really go to extremes.

I have a collection of 24/96 and even 24/192 flac files but its not material I listen to, much.  it ends up being 'demo material' more often than not.

the regular stuff I listen to does not need high-end playback gear.

but then again, if we build our own, its not expensive, it uses better parts than a sony (etc) would put in and we can change things if we want to.  I built audio gear because its not expensive, I can do it and I can get much better specs than most mass producers are willing to spend on.  but I'm not saying I hear the extra difference.