Computer Audiophile Frequently Asked Questions (and Answers)

#1
I've been a member of a local audio club on and off for more than twenty years. These are wonderful places to experience new gear, make new friends, and discover great music. As the years have passed, the typical playback source for club meetings has gradually transitioned from vinyl and open real tape to CD/SACD, and now to a PC or tablet transport and USB DAC. Whenever we were hosting a program with a PC transport source, the presenters would often be pounded with questions from the audience about computer audio since even as late as 2014 this was a completely new area for the majority of club members.

To help cut down on the number of questions asked during the meetings and therefore have more time for listening and fellowship, I typed up a Computer Audiophile Frequently Asked Questions (and Answers) document. While I have not yet finished the document, I thought I'd share a link here for members of Audio Haven who are just getting started with computer audio. I can't promise that all of my answers are 100% correct--most are fairly well researched while others are just my personal opinion. Still, I hope some find it helpful. Comments, suggestions, and corrections welcome. :-)

This forum topic may also be used for posting questions that I've not thought of (which I'll try to answer with help from the many knowledgeable folks here). Enjoy.
 

marantzfan

Administrator
Staff member
#3
Thank you for such a great source of information for those getting into digital. Looks like our first sticky in the Digital forum.
 

dewdude

Junior Member
#7
Oh boy...the upsampling section. I have a few comments on that:

There are a number of theories for this, but it could be because more gradual low-pass filters can be used during playback.
It's actually easier to design a good filter for a much higher rate where your errors can be pushed in to in-audible ranges. This is part of the reason behind oversampling...at least before it gets in the DAC. There's also other reasons such as DACs being locked to one rate...being optimized for a specific rate...or to get around any re-sampling your OS might impose on the audio stream. One infamous example were the Sound Blaster cards. Most of them that included a DSP ran the card at a solid 48khz regardless of what you fed it; and it had hardware SSRC to handle that. It was a very lousy SSRC engine that put a lot of IMD in to the mix. That was one of the reasons some software players included internal SSRC engines; they could do a slightly better job of going up to the required 48khz rate the SB cards wanted. Another imfamous example was the SB Audigy 2; while it had 24/96 capability; it still had a 48khz locked DSP and the only way to get "clean" playback out it was using special drivers that not only let you disable the DSP chip entirely; but pipe your audio directly to the "p16v" module which handled HD decoding. With the DSP turned on; your 96khz stream got resampled to 48khz for DSP and mixed with the 96khz output; and with the DSP turned off, your 44.1 still got upsampled to 48khz.

So part of the idea of upsampling in the PC playback world came from those limitations. Most DACs these days do not have such a limitation; and largely what you're trying to do is match the sample rate of the mixer in your OS.

At the same time, DSD playback only requires a simple LPF on it's output to cut the super-sonic noise caused by the format. OF course, 1-bit formats have a major issue with dynamic range; even at DSD rates...the higher in frequency you go, the less dynamic range you have due to required noise shaping.

Of course, DSD and PCM are like Apples and Lemons....both are fruit and grow on trees..but that's where the similarities end.

The other thing to bring up is the fact that, at least last I looked, any DAC carrying more than a 16-bit output was going to be using an oversampling method internally; while DACs with 16-bit output had maybe a 50/50 chance of being direct PCM or oversample. Oversampling at the DAC stage has been around for years; and back in the 80's was the only acceptable way to get a decode of 16-bit audio. PCM data is converted to a high-rate 1-bit stream; and then that 1-bit stream is converted to audio...with the same noise-shaping limitations that goes along with DSD (as that's literally a limitation to all 1-bit audio formats). Whether it was called straight oversampling, MASH, 1-bit dual D/A converters...or that Pulse Width Modulation-long-thing-JVC-called-it....it was the same principal.

One other thing to consider in the PCM vs DSD arena is that DSD *usually* has fewer steps to go to analog; you're already dealing with a 1-bit audio stream, so you feed it to a 1-bit decoder; there's no having to convert the PCM to 1-bit...so if you're all about purity in your audio path; PCM generally has addtional steps to it that DSD does not. On the flipside of that coin; a lot of SACD players actually convert to PCM before decoding (which is wasteful); and most of the SACDs on the market aren't pure DSD source to begin with. Why? You cannot edit 1-bit audio the same manner you can PCM...so just about every SACD on the market went through a phase of insanely high-rate PCM for editing/mastering before being converted back to DSD for the disc. In fact, I've seen some Japanese SACD releases that were sourced from 24/96 files...they said so in the booklet.

I personally don't think anyone should be storing upsampled copies of music..it's just wasteful. It's much more efficient to find a player with a built-in good SSRC engine. If the extra processing causes noise in your system; well there's a problem with your system...CPU usage and stuff like that shouldn't be causing noticeable noise.

No one should be storing in AIFF or WAV, ever, period. There is no advantage to using WAV over FLAC; if anything, you have disadvantages since WAV has no standard tag format and no error correction/detection. Should you somehow have a WAV file that's a bit corrupted, you will wind up with this annoying blast of noise..or worse...your DAC will lose sync and cause nothing but noise. At least a FLAC file will play to the corrupted part and throw an error, saving your ears...and your speakers. Again...if the CPU usage and what not does cause a change in quality; then you should be highly suspect of your USB host and PC in general.

Just my opinion on that part of the subject.
 

Shelby1420

Super Moderator
Staff member
#10
dewdude;n13009 said:
Oh boy...the upsampling section. I have a few comments on that:



It's actually easier to design a good filter for a much higher rate where your errors can be pushed in to in-audible ranges. This is part of the reason behind oversampling...at least before it gets in the DAC. There's also other reasons such as DACs being locked to one rate...being optimized for a specific rate...or to get around any re-sampling your OS might impose on the audio stream. One infamous example were the Sound Blaster cards. Most of them that included a DSP ran the card at a solid 48khz regardless of what you fed it; and it had hardware SSRC to handle that. It was a very lousy SSRC engine that put a lot of IMD in to the mix. That was one of the reasons some software players included internal SSRC engines; they could do a slightly better job of going up to the required 48khz rate the SB cards wanted. Another imfamous example was the SB Audigy 2; while it had 24/96 capability; it still had a 48khz locked DSP and the only way to get "clean" playback out it was using special drivers that not only let you disable the DSP chip entirely; but pipe your audio directly to the "p16v" module which handled HD decoding. With the DSP turned on; your 96khz stream got resampled to 48khz for DSP and mixed with the 96khz output; and with the DSP turned off, your 44.1 still got upsampled to 48khz.

So part of the idea of upsampling in the PC playback world came from those limitations. Most DACs these days do not have such a limitation; and largely what you're trying to do is match the sample rate of the mixer in your OS.

At the same time, DSD playback only requires a simple LPF on it's output to cut the super-sonic noise caused by the format. OF course, 1-bit formats have a major issue with dynamic range; even at DSD rates...the higher in frequency you go, the less dynamic range you have due to required noise shaping.

Of course, DSD and PCM are like Apples and Lemons....both are fruit and grow on trees..but that's where the similarities end.

The other thing to bring up is the fact that, at least last I looked, any DAC carrying more than a 16-bit output was going to be using an oversampling method internally; while DACs with 16-bit output had maybe a 50/50 chance of being direct PCM or oversample. Oversampling at the DAC stage has been around for years; and back in the 80's was the only acceptable way to get a decode of 16-bit audio. PCM data is converted to a high-rate 1-bit stream; and then that 1-bit stream is converted to audio...with the same noise-shaping limitations that goes along with DSD (as that's literally a limitation to all 1-bit audio formats). Whether it was called straight oversampling, MASH, 1-bit dual D/A converters...or that Pulse Width Modulation-long-thing-JVC-called-it....it was the same principal.

One other thing to consider in the PCM vs DSD arena is that DSD *usually* has fewer steps to go to analog; you're already dealing with a 1-bit audio stream, so you feed it to a 1-bit decoder; there's no having to convert the PCM to 1-bit...so if you're all about purity in your audio path; PCM generally has addtional steps to it that DSD does not. On the flipside of that coin; a lot of SACD players actually convert to PCM before decoding (which is wasteful); and most of the SACDs on the market aren't pure DSD source to begin with. Why? You cannot edit 1-bit audio the same manner you can PCM...so just about every SACD on the market went through a phase of insanely high-rate PCM for editing/mastering before being converted back to DSD for the disc. In fact, I've seen some Japanese SACD releases that were sourced from 24/96 files...they said so in the booklet.

I personally don't think anyone should be storing upsampled copies of music..it's just wasteful. It's much more efficient to find a player with a built-in good SSRC engine. If the extra processing causes noise in your system; well there's a problem with your system...CPU usage and stuff like that shouldn't be causing noticeable noise.

No one should be storing in AIFF or WAV, ever, period. There is no advantage to using WAV over FLAC; if anything, you have disadvantages since WAV has no standard tag format and no error correction/detection. Should you somehow have a WAV file that's a bit corrupted, you will wind up with this annoying blast of noise..or worse...your DAC will lose sync and cause nothing but noise. At least a FLAC file will play to the corrupted part and throw an error, saving your ears...and your speakers. Again...if the CPU usage and what not does cause a change in quality; then you should be highly suspect of your USB host and PC in general.

Just my opinion on that part of the subject.
Sooooooooooooooooooo much of this is way over my head, just now dipping my toes into the digital pool. I have tons of questions from reading this, first one is what does the filter do??
 
#11
Umm...yeah. I'm in the same boat as Shelby1420. I don't think I stand a snow ball's chance in hell figuring this stuff out. I think I need a digital front end "consultant".

Oh, and welcome to the Haven! Very nice to have you here!!
 

dewdude

Junior Member
#16
When talking filters in a DAC, one needs to remember your DAC itself won't produce a smooth output, it will produce a stepped waveform. The filters are there to smooth it out to something a bit more analog like. I've never studied the world of straight PCM dacs (dacs that decode to analog straight from PCM using a resistor ladder rather than 1-bit delta-sigma); they were always complex expensive devices with not much advantage over an oversampling DAC with the exception of not having an additional D-S conversion step.

I guess part of what helps if if you understand how PCM stores it's audio..it's essentially mapping amplitude levels of a waveform. Each sample is a representation of the voltage of an analog signal at that given point; represented as so many bits per sample. The more samples you have, the higher the frequency response...the more bits you have, the lower the amplitude you can represent before "running out of floor"...it also gives you a little more accuracy in the upper end...but with digital audio, it's the lower end of the scale that you really look at. If you have a track that only has a dymanic range that never drops below -6, then you don't need 24-bit to accurately represent that...it's only when you start going in to the real low amplitude levels that your range matters. So, for example, if your audio level was reduced to be an average of -90dBFS, then anything that drops below -94 or 95dB is essentially lost...there aren't enough bits to represent an amplitude that small. IT's why it used to be important to normalize your audio file in the editing stage to prevent quanitzation noise between steps. Basically, there are more bits required for lower amplitudes than higher amplitudes. There's also the old myth of "24-bit doesn't clip"....no, it will. This comes again from the world of audio editing, which is where most people encountered anything more than 16-bit audio. People skipped the fact we are editing/processing using floating point math vs integer data. Floating point allows for a much larger dynamic range than it's integer counterpart; my editor can represent a level of something like +20dbFS over clip before the edited file actually clips. You need that kind of range and overhead in editing though.

Anyway...back to the original point. PCM is basically like a game of connect the dots using a grid. Your vertical scale would represent voltage levels represented by the bits of the sample, the horizontal scale would be how many samples you have. This is how PCM stores it's audio...it's "positional" as in the sample contains all the information needed to map it to a voltage output. Do this 44,100 times a second and you get audio...but it's not exactly clean; that's where the filters come in. They'll remove some of the ultra-sonic junk and "smooth" the waveform out. As I may have (or should have) mentioned; I don't really know much about direct PCM DACs...ones that don't use an oversampling technique. I switched to 24-bit DACs 12-years ago when they first came out...not to mention most study revolved around CD players...and the majority of them used the oversampling technique.

So...let's move on to 1-bit/DSD formats. These do not actually store any samples...after all, you're only dealing with single bits. So how does that all work? 1-bit stores the *changes* involved to make the analog waveform, rather than storing the actual voltage values. You have a sawtooth wave generator that's capable of running at your max sample rate...so for DSD it's 2.882mhz per channel...that's a really fast generator and ultra-sonic signal. But it doesn't run automatically at 2.8mhz, it's just capeable of being flipped at that rate. The bits in a 1-bit format basically control the sawtooth generator. Send it a 1, it changes direction of the sawtooth generator; send a 0, it continues as normal. By controlling how often this thing changes direction; you can ramp up or drop the output voltage accordingly. Of course, it's changing faster than the actual audio...so you have to pulse it to keep it in the approximate area. Surely you know what a sine wave looks like...now picture there were lines above and below that sine wave...now draw a very rapid sawtooth between those lines? See what you get? You get an ultrasonic saw-tooth that "follows" the analog output. You don't hear the very rapid voltage changes becuase they're way above what your hearing can handle; but you are able to pick up on the larger changes in voltage. All you really have to do...is apply a low-pass filter to he audio to get rid of that. There's also distortion from noise-shaping...which the LPF also helps to rid of.

The 1-bit method is actually a lot easier to generate analog from vs having to have a component for each bit and then sum them together. It's not perfect, it has it's limitations...just as analog audio does. And what I gave is a very over-simplified version of it. I don't have a good technical understanding I can fully explain...just the "bare bones differences" between the two. If you really want to be confused, you can read https://en.wikipedia.org/wiki/Delta-sigma_modulation on the finer details of delta-sigma modulation and https://en.wikipedia.org/wiki/Digital-to-analog_converter#DAC_types will explain, in moderately more simplistic terms (but not as simple as I attempted to do) the differences and issues with each type of DAC on the market.

I mean..digital is complex...but I think people are making it out to be more complex. One thing that bugged me is the number of things people would claim have an effect on digital...most of which aren't related to the actual medium itself. When I hear people claim that FLAC is "still compression" and that "the extra CPU cycles hurt quality"...I actually want to go in to a rage because it shows a total lack of understanding of the big picture. People that make claims like that usually have a nice DAC but inferior components in the PC...a cheap USB host can require more CPU overhead...which can cause buffering issues and glitches. A cheap motherboard will have all sorts of noise on it's power supply that leaks in. There's a reason I use a specific laptop for audio playback; it contains a USB host with almost zero CPU overhead required...and I don't start getting quality issues till I'm doing something like mixing 32 tracks wit 66 effects...and at that point the CPU just can't render that properly. It's like I've said...if components like your hard drive, ram, or CPU usage cause quality issues...you really need to look at your motherboard than your hard drive, ram, or how you're playing the music back.

But that's all just my $.02 on the subject. I've been messing around with digital audio since long before I was ever interested in analog performance.
 

billfort

Administrator
Staff member
#17
dewdude;n13329 said:
I mean..digital is complex...but I think people are making it out to be more complex. One thing that bugged me is the number of things people would claim have an effect on digital...most of which aren't related to the actual medium itself. When I hear people claim that FLAC is "still compression" and that "the extra CPU cycles hurt quality"...I actually want to go in to a rage because it shows a total lack of understanding of the big picture.

But that's all just my $.02 on the subject. I've been messing around with digital audio since long before I was ever interested in analog performance.
Yes, I've noticed in the past that can get you into a rage at times on this stuff Dew. :)

And I'm glad we are seeing the acknowledgement that personal opinion sometimes creeps into a great lists of 'facts' like this. Some (guilty! as someone who has been messing with analog long before being interested in digital :D ) believe they 'hear' differences in things like flac vs. wav, computer implementations and power supply quality - even though the numbers and theory say no-way. Numbers and theory even said digital beat analog from the very beginning, but it certainly didn't sound that way to me. I'm the first to admit however, that the evolution of the digital art has completely changed my view, and I now embrace digital right along with vinyl on it's sonic capability. I personally think format, filter design, processor implementation, DAC chip design, compression (or lack there-of), network implementation, power-supplies, etc. had and have an important role to play in that evolution. I think there is a lot more to this than 'bits-is-bits' and it either works or it's broke - that's what my ears tell me, just like they told me digital wasn't all that back in the 80s when the numbers where used to 'prove' it was all perfect.

The cool thing about this forum is that we all get to contribute and voice our opinions whatever they happen to be without being 'told' you are full of it - right or wrong is in the ears or analytic mind of the beholder. I won't go into a rage if you say computers don't matter and please don't go into a rage of I say I prefer wav to flac - deal? :)
 

dewdude

Junior Member
#18
At least with digital we can go back and accurately meausre differences. The FLAC vs WAV thing for example; I tell people to convert that FLAC to a WAV and tell me if they aren't bit-for-bit exact; if there were any differences, no matter how subtle; the file would not come out bit-for-bit matching the original.

As far as "network implementation"...that shouldn't have a single thing to do with the quality. You are still transferring the same bits regardless if it's WiFi or ethernet. What mostly comes in to play with digital is the design of your DAC (including it's method of going analog and filter...on top of the construction of the analog stage of the DAC), power supply..how you're driving the dac (getting audio in or around OS mixers)...as well as what you're putting in to it. Everything else, in my opinion, is an analog person reaching for problems. They see problems, they see a challenge to fix. It's like with ham radio...some guys feel as if they're not constantly tweaking something they're not having fun.

Early digital was pretty rough; we didn't have ADCs capable of true 16-bit and DACs were far from perfect...there's a LOT to be desired even though the numbers said this....it's a case of the hardware hadn't caught up. But it's been close to 50 years they've been doing forms of digital audio...maybe the last 35 when they've been focusing on audio performance (the first digital audio circuits were optimized for telephony). It's like why do early MFSL discs sound much better than any of the counterparts? Mostly because they used an analog-digital converter that was based on ICBM navigation stuff...you actually needed a security clearance to even look at it.

I'm actually not saying "computers don't matter"...I'm saying people tend to focus on the wrong aspect of the computer. They focus more on things like cables, DACs, hard drives, ram....when the only one of those that has a major effect is the DAC. Your computer does matter, it matters a lot...you just have to know what to look at. The quality of the rest of your PC matters more than what format you play back. I just...I don't understand why guys prefer WAV to FLAC. If there was a logical reason, I'd be ok...but the guys that do so tend to do so because of some kind of misinformation or assumption. To me, saying WAV is better than FLAC is like saying global warming/climate change doesn't exist because it was cold yesterday.
 

Shelby1420

Super Moderator
Staff member
#19
Fascinating stuff!!! Good to hear about computer importance, I built this computer with music in mind, every piece of it has THAT first and foremost!!! I will need to read this over and research this to figure out what half of it means, but after listening to Bill's digital system, it is worth it IMHO!!! Thank you!!
 
#20
dewdude - Whatever the measurements may show, I trust my ultimate measuring tools - my ears.

That's the beauty of The Haven: We can all discuss our experiences, observations, conclusions, etc. without arguing about things or being told that we can't possibly hear what we know we hear.

YMMV..... :)
 
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