Massively improve the SQ of computer audio streaming?

[beerandmusic]

INRE>>> "A novel way to massively improve the SQ of computer audio streaming" thread.

 

i tend to stay away from that thread most of the time....it's too big, too many rules...pretty soon it will turn into it's own website.

I would much rather see threads separate from it...like "best audio streamer" rather than that jargon get lost in that forest....i know, it's got a table of contents...LOL.  I do occasionally look at it, to see if there is anything new...just found out about audio-linux, but if audio-linux had it's own thread, it wouldn't get lost in that maze and I likely would have found it months ago.  People are too prudish to "outsiders" in there too and don't like to help....let that clan have it.

 

Besides, it's like every other page, it's LIKE OMG, best SQ ever!  I can't believe my ears....I mean, really, how many times can that really happen?  I will wait til everyone agrees in that thread that there is a concensus, and skip about 5000 pages, and jump on that solution.....right now it looks like it's audio-linux...what will be the OMG next month.

 

[beerandmusic]

1 minute ago, lmitche said:

I agree with most of what is written here, but this. As we sort out software, power sources and learn more about cable shielding and the instrinisic noise of various storage devices and  clocks, etc . . . we are starting to see increasing returns, not little subjective impact.

it started with USB toys about 7 years ago on this site....i am ready for a $500 streamer dac that is even 10% better than listening to vinyl on jbl speakers and audio research tube amps 50 years ago.

[beerandmusic]

9 minutes ago, fas42 said:

 

I'm in accord with you - because I count "software, power sources and learn more about cable shielding and the instrinisic noise of various storage devices and  clocks" as elements of the 'network' of poor implementation - a robust solution would have have all of that sorted, and from then on the returns of other changes are far less impacting.

 

the altair had LPS, network, ultra low noise clocks, linux about 3 years ago....if anything beats it, it's just subjective...

 

[beerandmusic]

2 hours ago, One and a half said:

By adding a component into the signal chain from a computer to a DAC for instance, even if it uses power from the USB bus is introducing capacitively coupled noise into the system.

 

 

this is why a streamer dac like the altair in design seems to make sense....if designed right and at the right price point.

less cables, interconnects, etc...but if they get the streamer to a reasonable price (e.g. audio-linux for $30 or volumio FREE), then I am all for a dac and streamer as 2 different devices.  I never understood the crazy prices, when one solution is only subjectively better, and not for all genres, all instruments, all songs, etc....e.g. cymbals sound better on this solution and bass sounds better on that solution.

 

[beerandmusic]

just because i think any one reading this thread, should consider if they can't budget an sotm ultra... ( a relatively low priced streamer and decrapifier in one unit).....the PRO-JECT STREAM BOX S2 ULTRA

 

 

 

[beerandmusic]

9 minutes ago, jabbr said:

My linux sounds better ... its all in the latency with which my fingers click the kernel compile options

 

Seriously ... there is effectively one set of realtime kernel patches that any and everyone can use ... also the option of running in RAM is available to everyone ... what is great about Linux is that it is open source. 

 

Its great that audiolinux has built and packaged a version of Linux suitable for end users! I encourage anyone who is not inclined to built their own kernel, or optimize their own package. There are so many options and it takes work to package and maintain. For folks who just wish to use NAA: @Miska's prebuilt NAA images are excellent (he has also optimized)

 

In any case: Linux is ideal to optimize as a stripped down lean and mean audio machine because you have total control over what goes into the box. You have access to the source and can modify to suit whatever needs you have: audiolinux is a perfect example!

 

Sounds greek to me, but i trust you (lol).  I have a question for you..is there any way to know if the linux kernel in volumio is comparable in SQ or not?  It has 2 features i like better than audio-linux (FREE and Web browser player)

 

https://volumio.org/project/

or

https://volumio.org/

[beerandmusic]

2 minutes ago, mansr said:

That's another nonsense claim. Every modern system executes code residing in RAM. It is impossible for the CPU to do anything else.

i think they mean entire os running in ram so it doesn't need to retrieve parts not in ram first.  I could see that as a benefit.

[beerandmusic]

9 minutes ago, jabbr said:

 

I don't know the details between the two.

 

Most all linux kernels are the same except for compile time options but they share a common low level audio library and drivers (ALSA)

 

The differences are mainly what additional user software packages are included.

 

so the SQ should be close to same, regardless of which linux kernel (linux-audio or volumio), and would more depend on the player software (assuming same hardware)?

[beerandmusic]

8 minutes ago, mansr said:

If you have enough RAM, everything ends up cached there anyway.

some os's specifically windows will only cache some portions of the os once it knows it is required, regardless how much free ram it has...again, i can see benefit of loading entire os into ram and never having to cache parts of the os in.

[beerandmusic]

21 minutes ago, jabbr said:

 

I don't know the details between the two.

 

Most all linux kernels are the same except for compile time options but they share a common low level audio library and drivers (ALSA)

 

The differences are mainly what additional user software packages are included.

 

I think I am just realizing main diff between volumio and audio-linux.

 

audio-linux allows different apps to run on it, where volumio linux kernel only allows volumio player (at least without hacking it)...

 

Maybe it will make sense for me to have 2 linux boxes, one for everyday that i can control from web browser, and another that i can fiddle with and use different players.  Either way, volumio is free and supports web control, so i think i will try that one first.  I have been wanting a web based player forever that supports DSD...and to think its free and will run on just about any hardware you got laying around.

[beerandmusic]

6 minutes ago, Miska said:

 

USB has constant packet rate. So does HQPlayer - NAA link.

 

When you use smaller buffer you increase processing overhead, so you are wasting energy. It also moves the packet frequency up, making the possible noise more audible. Since transfer is usually DMA based, it is very efficient. So larger the buffer, less electrical noise you have. This overhead is also reason why gigabit and faster Ethernet usually support so called jumbo-frames, bigger MTU.

 

Since you are here, i have a question...if i have hqplayer license, can i use that license for embedded?  I am thinking of trying this audio-linux with hqplayer embedded?  Also someone said they thought i can control the hqplayer with a web browser if i use the embedded version?

[beerandmusic]

43 minutes ago, Miska said:

 

Desktop and Embedded have separate licenses... But Embedded has same kind of trial as Desktop, so you can always try it out.

 

Yes, with Embedded you can play albums from local library using web interface. Currently it is very bare bones, but I will add more functionality to the web interface over time...

 

very bare bones?  all i need is to be able select track, volume, and skip?  Is there a picture of the interface in the documentation or anywhere? thanks...i googled but couldnt find anything about web interface?

also, besides having web interface available, Is there any other benefit to embedded over desktop for home use, if i am going to control it remotely (either via web or HQPDcontrol)

[beerandmusic]

1 hour ago, esldude said:

This part of my skepticism.  Multi-factor analysis would show that something like a computer is very complex.  The chances these processes make the a big difference in sound quality downstream which would be heavily filtered and not directly dependent on any one of them anyway seems like grabbing straws in a haystack.  The whole irradiated interference just doesn't seem to pass muster.  

it seems logical to me in that we are talking high resolution and timing is everything in processing ....i am just surprised that it is taking so long to "perfect" these obvious things.

the thing that keeps baffling me is that everyone seems to agree that the dacs get the bits perfectly. yet there is apprarently so much difference in SQ, and yet it is not measurable, and just because someone likes one kit better than another, doesn't even indicate if it is more accurate or not....if everyone agrees the dacs get the bits accurately, then is it just about who has the better "noise"?  I just am not so convinced that dacs get the bits with 100% accuracy?

[beerandmusic]

inre latency...not sure it is relevant or not ....from jcat:

2. Adjust latency in USB DAC’s control panel

This is one of the things that have the biggest effect on sound quality in a computer audio setup. Not all DAC drivers come with a buffer size/latency adjustment, but if yours does, I highly recommend setting it to the lowest value. The values are often shown in milliseconds, but can be also presented descriptively; e.g. normal, safe, minimal. You want the 1ms or the minimum latency setting. It will sound smoother, more dynamic, more detailed and with a better control.

 

[beerandmusic]

8 hours ago, Superdad said:

 

You are still in denial after all these years Dennis?  The differences people report—and which I plainly hear every day in my system—are far from random.

It does take a bit of training and comparison to tune your ear/brain into the type of sonic difference signal integrity improvements make.  Rather similar (though often more dramatic than) the variations heard between different digital filters.

Oh wait, I seem to recall reading you say you don’t hear differences between digital filters either.  Sorry...

 

So are you on the bandwagon amongst the vast majority that believe that dacs get their the music with 100% accuracy?  And if so, and there are clearly notable SQ differences, then it is just a difference of noise then?  Basically we are paying big bucks for who has the "better noise" and that is the entire difference (up to the dac)?  After all, all that is being transferred to the DAC is the music with noise and voltage with noise??

 

and if so, this continues to baffles me...is engineering so stupid after all these years that they can't get it "right" or at least within 5% of the cheapest to most expensive solutions (at least up to the dac).

[beerandmusic]

8 hours ago, Superdad said:

I think people respond differently to some changes—and some choose what I would consider the worse and less accurate choice

 

 

This i agree 100% with and have thought it a LOT.  i think a lot of people that report OMG improvements are actually liking something less accurate....this has to be true, because so many people report OMG improvements, and you can really have only so many OMG improvements.  This is another reason I am a lot more OBJECTIVE than SUBJECTIVE, and not willing to pay for "subjective" improvements...they likely are not even more accurate.

[beerandmusic]

8 hours ago, Miska said:

 

I'm personally using 100 ms buffers and that is certainly good compromise. That keeps buffer rate at 10 Hz which is below audible frequencies. When you push it to 1 ms you also push the buffer rate to 1 kHz which is within audible frequencies, and in addition increase overhead and produce more noise. USB has 8 kHz packet rate (packet size depends on sample rate), and that frequently shows up in measurements leaking to analog output of the DAC...

 

I never noticed any difference to those stated qualities by changing buffer size.

 

For that same control panel, I set USB Streaming Mode to safest setting and ASIO Buffer to largest setting. Gives best reliability (least likely to have drop outs) and least computer activity.

 

+1

worth noting for all hqplayer enthusiasts.

[beerandmusic]

14 hours ago, jabbr said:

 

I find video to be very stressful on a computer and my first experience with really bad noise was when I was running 4K editing (a few years ago). Smooth, nonstuttering playback requires substantial resources and highlights what a cheap 4K blu-ray player can achieve. Working with video and having noisy and high powered graphics cards does impact SQ in my experience.

 

My solution is to use Ethernet to isolate the noisy PC from the DAC using a very low powered NAA running Linux without a GUI etc. Even with a really low powered Celeron or ARM device, the NAA hardly uses CPU.

 

Low power, low noise, sounds great!

 

This sounds logical to me.  I know HQPlayer has NAA, is that what you are referring to that you use?  That is one of the options that audio-linux offers.....I really don't know much about NAA, but it does sound like a good way to go.  would you need more than one computer in that configuration as the host?  e.g. can i have just one audio-linux pc running naa and point to local hd, and then have a separate pc (or smartphone) to control it, or would you need another pc or nas for the music files?  I am willing to go this route if just one pc (besides the controller).

[beerandmusic]

3 hours ago, jabbr said:

 

What I do:

 

1) music and all other media on a central networked NAS.

2) Workstation PC (or embedded server) with GTX1080 runs HQPlayer (and Roon) -- needs to be beefy to upsample to DSD512 etc.

3) several linux NAA boxes, either hardwired or wireless, connected to several DACs by USB. The NAA boxes aren't expensive -- I've described a number of mine including the ClearFog, Espressobin, Celeron J1900, NUCs, RaPis etc.

 

I use my iPhone to control Roon. Alternatively you could use an iPad with web interface to control HQPe. 

thanks.

 

1. why several naa boxes and several dacs instead of just naa box and one dac?

2. my  main question is can the music be on the same box as the naa? 

 

i reallly only want a 2 box solution (player and host or whatever you want to call it)..i suppose i could buy a nas later, but i really just want a 2 box solution.

 

edit: i guess black friday is here, i guess i can pick up a nas cheap these days....which do you recommend on the cheap?

 

i really have been staying away from nas as i am not sure what value they add over just a network share since i have many computers and ssd drives here already if i am not concerned with raid. (i use ssd exclusively and have an automatic backup system already).

[beerandmusic]

1 hour ago, lmitche said:

Let's expand on a framework that Jussi discussed somewhere else, the extreme case of DSD512. This is 22,579,920 bits per second. Now let's assume that we have a 1 core 4ghz processor that never waits for data so 4 billion instruction can execute a second. 

 

Dividing 4 Ghz by 22mbps is 181. This means that there are only 181 instructions that can be executed per bit in a dsd512 stream on our fictional 1 core processor. OK, so we have four cores, now we are at 724 instructions per bit. That's a very low budget for a realtime process especially one that is doing bitwise operations.

 

In this context, the impact the processor doing anything unrelated to audio, like the Windows OS that constantly plays with itself, one can quickly see that things can get stretched or delayed waiting for a processor.

 

Understanding this also tell us that what Jussi does in Hqplayer is a modern miracle. It also may explain why AL has such a great impact.

 

And yes, I know, the real world is much more complicated with lower resolution than dsd512, DMA, caches, efficient compilers, instruction look-ahead and execution . . . blah, blah, blah. 

 

Larry

 

Can you explain why if the bits get to the dac with 100% accuracy like everyone agrees, why any of this stuff matters?  I am not doubting, i just am looking for logical reasoning as to why?  This stuff sounds like it would affect timing and data accuracy,, but not sure how it would affect noise?

[beerandmusic]

9 minutes ago, lmitche said:

Nope, unless they arrive too late.

arriving late would affect data or noise?  everyone agrees that the dac gets the data with 100% accuracy, so arriving late affects "noise"? 

 

I am curious why everyone thinks that the dac gets the data with 100% accuracy (at the right time)?  is there any way to really determine that the dac gets the data with 100% accuracy at the "right time"?  I mean its at very high resolution and very fast....And if that is really the case, then the only thing it can be is "noise"?

 

Also, if dacs get their data with 100% accuracy at the right time, then why do low phase clocks even matter?

 

It seems to me that a solution should be perfected (at digital end, prior to analog out) easily if you had a dac engineer that designed both the digital in system including clocking/timing and the dac, rather than 2 different engineers trying to piece it together.  But it will take a pretty good dac engineer (smile).

 

I wonder if the front of a streamer dac (e.g. altair with lps, ultra low noise clocks and dac), if the digital front end is
"as good as" these other linux streamers?

 

[beerandmusic]

13 minutes ago, mansr said:

Late arriving data results in dropouts or pops/clicks.

 

Yes, i have heard that before, but I am not sure I am 100% convinced of that.  I believe it will cause that, but is it also possible that if the data is 100% accurate, but dac doesn't process at right time that it could cause something besides a dropout?  I mean if you have a million bits at t1 and the dac gets a million bits minus one bit at t1, and the millionth bit is carried over to t2 along with another 1 million bits, would you really hear a dropout?  Of course these are hypothetical numbers, but just saying.  I have to think that some of those bits are for harmonies where the loss of 1 bit may make the harmonic sound slightly different, but not a dropout.

 

Anyway, that is my thinking, and probably everyone and their mother will disagree with me, but i don't think anyone can convince me otherwise..i am pretty stubborn and ignorant with my logic if i can't see it plainly (grin).

 

I just find it very difficult to believe that we are just paying for who has better "noise", if the data is always 100% accurate gettting to the dac at the right  time.  Plus that no one can identify which "noise" is more accurate...yet there are so many "omg" moments.

[beerandmusic]

^^^more thoughts to myself....

 

If Miska could answer above, i am sure he would sound logical, and I may be able to accept it, that if even one bit of a million bits arrives late, it will always result in a dropout that could easily be heard, ....provided he could also explain why so many people report OMG improvements based just on noise, not the data (since the dac reportedly receives the data with 100% accuracy and the only other thing being transferred is noise), and that is why people report omg differences that this "noise" can't be managed at least to an audible level, that this noise can cause dramatic depth, soundstage, detail, bass, etc... differences that cannot be managed effectively.

[beerandmusic]

^^^^ I think i answered my own question...and i was half right (ok, call it 1% right-grin)....

 

the timing is IMPORTANT because the noise (jitter), does cause "harmonic distortion".  I knew the SQ concernt would be more along harmonics than dropouts, but I guess i must admit, it is more about jitter than accuracy.

 

++++++++++

https://www.design-reuse.com/articles/5763/timing-key-to-optimizing-audio-performance-in-consumer-products.html

 

most notably: As little as 1 to 2 nanoseconds of clock jitter can cause a large degradation in the system's ability to play back a wide range of audio content (dynamic range) and an increase o f harmonic distortion

++++++++++

 

So it really is about timing, probably more than anything...more than power (not to say it doesn't matter)...but when we are dealing with such high speed processing, we need to control the jitter.  I am not convinced that any front end with good clocks (coordination between dac and source), that does not have "unnecessary interrupts", should be very cheap and easy to do.

 

This probably answers why usb toys works too, because of pc's (most common method of usb dac in "passed").  THis is why i always preferred enet to usb even before usb toys, likely because of non-dedicated pc doing processing).  USB toys and even enet probably is not necessary, provided a front end with good clocks without unnecessary interrupts... i also am not sure reading from a local ssd would cause any more unnecessary interrupts than streaming from enet.

 

[beerandmusic]

^^^^ the fact that nano-seconds of jitter can cause dramatic differences to dynamic range and harmonic distortion, does show the importance of timing, but IMHO, with engineering what it is today (maybe not in audio world, but computer world), we should be at a point at this day and time to have this stuff nailed down.  I am unimpressed with audio engineers, that they cannot get this done cheaply a LONG time ago (i mean litterally decades ago).  this should not be rocket science, even if we are talking nano-seconds.   With today's computer power, clock circuitry, this should be rudimentary at best....and perhaps it already has been perfected cheaply (to an audible level), just marketing, sales, and ignorance, continues to work on the ignorant. 

 

Now that i have come to this realization, it is probably time to invest in a cheap front end and cheap dac and be done with all this non-sense....i am not even sure that "BIG BOX solutions" are not already there, but marketing, reviews, ignorance, and survivability of boutique audio shops, have refused to let it mature...at least for the front end.  I am flabbergasted that it has taken to 2018 to realize what we know today.

 

I think part of the reason, i dismissed "noise/jitter" as the sole reason for digital SQ, is because i have heard before that noise/jitter sounds like hiss (not that it can cause other issues as well, such as harmonic distortion).  So i will forget all those posts that say jitter sounds like hiss, or listen to this wav to hear what jitter sounds like)...jitter is a bigger animal than hiss.