Linux Audio the way to go!? - Page 105 - diyAudio
Go Back   Home > Forums > Source & Line > PC Based

PC Based Computer music servers, crossovers, and equalization

Please consider donating to help us continue to serve you.

Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
Reply
 
Thread Tools Search this Thread
Old 28th March 2009, 03:45 PM   #1041
diyAudio Member
 
soundcheck's Avatar
 
Join Date: Mar 2005
Location: D
Quote:
Originally posted by ewildgoose


but the point is that it's not a "PC issue", it's a "I need a more robust DAC" issue

If you read what I am writing all the time: 99% of all setups do work with "less robust DACs".
Improving this situation substantially, with no cost associated to it - i'd call a fair deal.

Cheers
  Reply With Quote
Old 28th March 2009, 05:58 PM   #1042
diyAudio Member
 
Join Date: Jan 2004
Location: London
This is a forum for people who "measure", not a forum for voodoo magic.

Please provide your evidence that either:

a) the software is mangling the "bits", or
b) the DAC is sounding different depending how you feed an identical set of bits to it

There is little point simply doing voodoo magic where clocking the processor 2/3 the speed sounds better or feeding the same bitstream with application A "sounds better" than using application B. Find the source of the problem and fix it rather than trying to wave a magic wand

Measure, measure, measure (not with your ears, use something objective)
  Reply With Quote
Old 28th March 2009, 06:17 PM   #1043
diyAudio Member
 
soundcheck's Avatar
 
Join Date: Mar 2005
Location: D
Quote:
Originally posted by ewildgoose
This is a forum for people who "measure", not a forum for voodoo magic.

Please provide your evidence that either:

a) the software is mangling the "bits", or
b) the DAC is sounding different depending how you feed an identical set of bits to it

There is little point simply doing voodoo magic where clocking the processor 2/3 the speed sounds better or feeding the same bitstream with application A "sounds better" than using application B. Find the source of the problem and fix it rather than trying to wave a magic wand

Measure, measure, measure (not with your ears, use something objective)

I expected something like that. If you guys run out of ideas or arguments, you just bring
up the "measurement" story.

I can't afford mega-buck equipment. Do you expect me to purchase a >20k analyser to proof that a 100$ DAC sounds 5% better with ecasound than MPD. On what planet
are you living?

Man - I just type a command and listen. That's about it - period.

I've seen many of these so called "gurus" who actually failed on delivering measurements from inside a PC. I havn't seen very much of these (relevant) measurments at all.
  Reply With Quote
Old 28th March 2009, 06:32 PM   #1044
anli is offline anli  Russian Federation
diyAudio Member
 
Join Date: Oct 2004
I'm agree with ewildgoose - if you hear difference, just fix bugs in used sw. I'm not too far from Linux audio software (being QLoud author), or from listening to music (being listening to jazz 24/7). Probably, I have tried almost all players/engines under Linux (at least those ones which are in the Gentoo portage) and have not noticed any formal or emotional differences during all these years (BTW, RME HDSP 9632 is in use). Of course, I'm not saying about cases when this or that sw has a bug.

Ok, I see, there is another opportunity - probably I'm deaf... Well, also have not noticed this fact

Well, there are some playing back problems in the other (m$) camp - every piece of sw audio chain tries to do something "smart" with a sound. But here (Linux) we live in lucky world, don't we?
  Reply With Quote
Old 28th March 2009, 08:30 PM   #1045
diyAudio Member
 
Join Date: Jan 2004
Location: London
Quote:
Originally posted by soundcheck



I expected something like that. If you guys run out of ideas or arguments, you just bring
up the "measurement" story.
This sounds like the old - the world is flat - I refuse to believe it or go find some evidence either way

Quote:
Originally posted by soundcheck


I can't afford mega-buck equipment. Do you expect me to purchase a >20k analyser to proof that a 100$ DAC sounds 5% better with ecasound than MPD. On what planet
are you living?

Man - I just type a command and listen. That's about it - period.
Which comes back to my other point - perhaps you are on the wrong forum. Quite clearly recording the digital output of a sound card by feeding it back into the digital input of (same or other) soundcard is kind of basic level for someone who is arguing this subject

I was actually expecting you to be at least a bit higher level and at least claim that they didn't match because there was random amounts of silence at the beginning or something, but failing to grasp how to record the digital output of your own card is well below what I was expecting...

There are plenty other ways you could capture the digital output. The above seems like the simplest, but I doubt too many here own 20K analysers either - this is a DIY forum... You are just being silly...

Quote:
Originally posted by soundcheck

I've seen many of these so called "gurus" who actually failed on delivering measurements from inside a PC. I havn't seen very much of these (relevant) measurments at all.
Oh well, I'm just one of those simpletons who has actually tried to test this... You can find my "rec_imp" application at http://www.duffroomcorrection.com - this sends an impulse to the output and records the input and if you do a loopback then you should get a dirac pulse if the loop is perfect. Curiously the offset of the pulse will also measure the loop delay from output to input (about 23 odd samples on my RME 9632 from memory).

It talks to quite a few output drivers so you can even measure (and see) that windows drivers sometimes mess up the signal instead of passing it through unaltered - also you can see when resampling is done badly and breaks thing

If you don't understand the above then at least go read up before responding - I hardly think it's sensible to be arguing voodoo magic if you lack even a baseline performance to objectively measure against.

If you measure then it's science - if you guess then it's just guesswork...
  Reply With Quote
Old 28th March 2009, 09:51 PM   #1046
diyAudio Member
 
Join Date: Aug 2008
Quote:
Originally posted by soundcheck

Man - I just type a command and listen. That's about it - period.
The problem is that listening is not an engineering process. There are no standard (measurable in any known units) for sound quality and vice versa. Additionally, sound quality perception is very personal and is subject of taste (and taste is not subject of discussion).

I think we should clearly distinguish personal listening impression from measurements with oscilloscope or any other industrial instrument. These are very different matters, and collision between them is useless waste of time.

Can two audio players sound different on the same PC system ??? If course they can because there might be some code built-in which modify/tweak audio source info for whatever reason.

But if they both send raw bits sound will be indistinguishable in blind tests.
  Reply With Quote
Old 28th March 2009, 10:48 PM   #1047
diyAudio Member
 
Join Date: Jan 2004
Location: London
Anyway, coming back to the core questions

- Jack seems to work reasonably reliably and has a fairly small overhead. However, it's network capabilities are a bit limited (cross platform). Also, it's not straightforward to dynamically start Jack at different sample rates (and startup a network of plugins at the same time, eg brutefir).

- It would be desirable to at least have the option to separate audio generation from the machine which consumes it.

- Pulse audio looks like the best future bet on an audio server which is cross platform, handles multiple output devices and has some ability to sync multiple remote network connected outputs.

- Anyone tried using pulse audio as the audio server with Jack as the output device?

I currently do everything on one machine so don't yet have a need for this - however, if I went dual box I would also like somthing like a Mac Mini as the audio server, which in turn means that I need to find a good firewire audio card which can do 8+ analogue outputs. The Echo Audio looks on paper to be one of the few supported devices which might work out. Anyone tried one of these?

http://www.echoaudio.com/Products/Fi...ire8/index.php

Any other ways to slice this one?
  Reply With Quote
Old 28th March 2009, 11:17 PM   #1048
EddieV is offline EddieV  Netherlands
diyAudio Member
 
Join Date: Aug 2007
Quote:
Originally posted by LinuksGuru

.... sound quality perception is very personal and is subject of taste ...

But if they both send raw bits sound will be indistinguishable in blind tests.

Hello LinuksGuru,

Both your quoted statements are wrong. The whole discussion above is for me a clear example of die-hard technicians against persons that are interested in music. In the 1970's I met many technicians that started to laugh very loud if I said that I could hear differences between cables. In the early 1980's lots of technicians told us that CD's are perfect.

Now we encounter technicians that do not hear the difference between one player and another. If you have lousy equipment and you are not interested in the details of the music then you can claim this. For me such a statement has no value.

I know the complexities of the sound of high quality acoustical instruments. I can understand why musicians spend a fortune on an instrument that they like. My goal is to get as much as possible of this sound from my music system within a limited budget. And I do hear lots of differences between players on my PC. Even with the same player used in a graphics terminal or in recovery mode.

I would like to know the technical/scientific reasons for these differences. But in the mean time I am very happy with the music that I can enjoy every evening. Almost three months ago I got the opportunity to swap Brutefir (brutus) against Ecasound (ecasx), (thanks to Soundcheck !). A clear example of how sound quality can be improved at no cost.

Someone else in this thread said "I do not trust my ears". Well, if you do not trust them, what do you trust?


Kind regards,
Eddie
  Reply With Quote
Old 28th March 2009, 11:35 PM   #1049
diyAudio Member
 
Join Date: Jan 2004
Location: London
I think he just misphrased his statement. Remember that he is talking about a situation with the *same* sound card being supplied *exactly the same* bitstream, only the name of the process supplying the data is changed - this seems to leave only very small reasons for the sound to be different in each case

Actually I *do* agree that it *could* sound different. Early in this thread someone pointed out that many cards have only mediocre PSRR and it's entirely possible that one program might inadvertently cause more RFI from the processor than some other program

However, my point was that science (not guesswork) could determine exactly where the difference was coming from. After all it's also possible that one program is mangling the audio data - if this is so then the two programs mentioned are opensource and can be fixed. On the other hand if the bitstream is measured and found to be exact then the audio card can be examined and sensible science applied to find the problem.

The problem with the voodoo approach is that you are trying to patch up one unmeasurable effect with another one. For example if it's some kind of powersupply noise getting into the card then the voodoo approach is to try loads of software players and pick one which apparently works best. The science approach is to improve the quality of the PSU...

The problem with the voodoo approach is that your results will perhaps not apply to me because my equipment may have different weaknesses, or be stimulated differently. The benefit of the science approach and fixing the PSU works everywhere...

Don't get me wrong - a few years back I used to dabble with cables and I certainly believe I can hear differences between them. However, now I view that as a problem with the equipment being "broken" if it doesn't like a certain cable and prefer to try and fix the equipment rather than go down the route of endless swapping of cables and other voodoo...

By the way - when you say you swap Brutefir for Ecasound - how are you doing your convolution if you dropped brutefir? Do you use some kind of plugin filter? If this is explained in another post then please just send the link?

Good luck
  Reply With Quote
Old 28th March 2009, 11:55 PM   #1050
diyAudio Member
 
Join Date: Jan 2004
Location: London
By the way the "I don't trust my ears" you might be taking a bit literally

I agree that some stuff sounds "better" to me. However, it can be for all kinds of reasons including what mood I am in.

I guess I have two competing goals. Sometimes I just want to enjoy some music. At other times I am trying to build the "best" system that I can. In the case of the latter it's helpful to use objective and repeatable benchmarks as far as possible

However, definitely "ears" are very subjective and I have definitely fooled myself on many occasions. Amazing even how a few tenths of a dB can distort the perception of which equipment is "best". I would definitely recommend trying some ABX testing on any stuff you are "sure about". It's quite amazing how the "clear" differences tend to evaporate (sometimes)!


Good luck

Ed
  Reply With Quote

Reply


Hide this!Advertise here!
Thread Tools Search this Thread
Search this Thread:

Advanced Search

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are Off
Pingbacks are Off
Refbacks are Off



New To Site? Need Help?

All times are GMT. The time now is 03:21 PM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2014 DragonByte Technologies Ltd.
Copyright 1999-2014 diyAudio

Content Relevant URLs by vBSEO 3.3.2