Huge audio improvement for free?

Just a couple class A's.  Followed one design from a book as I learned, and then modded out the second a bit.  Also modded a valve-jr. I use them for pure tube recording sounds, not powerful enough or versatile enough for live playing.

Sorry mate, I need glasses I guess, or a shrink for being inclined to read things in a negative way...

My sincere apologies!

It's cool, it happens, I'm more often a dick than not.

I had a serious question in there about whether at this stage building a high end tube hifi amp is worth it, or is loss-less digital enough for 99% of users using whatever card. I can run it through my digirack, or even fire up the old blackface adat or silverface for 24 bit pre's. Our band has in the past run that through tape just to "warm it up".

I've always understood that uncompressing audio is not possible (even from the audio engineers at the studio).

For some reason SoX doesn't show up in Audacious. I installed Audacious, its plugins, and then installed sox, (I think all its libraries), but I'm couldn't find a soxr. I found a libsoxr, and installed a few other libs but it still doesn't show up. Am I missing something? I also couldn't find the source code for this stuff, right now I'm just using Linux Mint repositories so that may be my problem.

Probably, you can find the full SoX project here: http://sourceforge.net/p/sox/code/ci/master/tree/.

The audacious project probably also has just the resampler plug-in packaged separately, I would go for that, the full SoX project is very nice software, but overkill for audacious, although for instance it could be useful if you want to use something like a maximizer effect if you play back music on small speakers for instance, that would be possible through the easy to use LADSPA slot in SoX, whereas most distros are on PulseAudio, and LADSPA can only be JACK'ed into ALSA, so SoX would offer a convenient solution there.

I tried the builtin resanpler and I'm not exactly sure if it's working or not, it still says it's playing my music at 44kHz so I'd assume not. Granted I haven't looked around the Internet for a solution yet so maybe it's an easy fix.

Cursory google searching has netted me that my audio card may not support higher sampling...*sigh* I'll keep looking but I just may have to deal with it.

I'm not sure about this.

Your numbers from the opening post (the effect-parameters) are not exact, are they? Comparing an original wav of one of my songs to the mp3-version (in banshee without any effects) and the mp3-version in Audacious (without the SoX resampler) gives me the impression, that the effects are just enhancers (maybe with nice techniques under the hood) and (stereo-)expanders that aren't that well suited for compensating the mp3-compression-loss.

The Audacious-version is somewhat more off from the original than the "normal"-mp3-version.

I'm not sure about the upsampling and the higher bit-depth too - even if the effects are using this headroom i would not expect to see much action in this area. And why  44,100 --> 96000? Why not i.e. 88,200?

 

Nevertheless the software seems to be very nice. I have to play around with this sometime. 

My goal is not the best-sounding setup (as an HiFi-guy would want) but a true (or at least well-known) setup - so i will stay with the non-compensating-version.

 

edit:

btw. I am one of those who would say, that you can't hear a difference between a 320mp3 and the original version.

Nice thread though :D

ok, iv installed windows version of audacious. iv made the tweeks bar the sox add-on. here the page:

http://sourceforge.net/projects/sox/files/sox/14.4.1/

witch one do i download?

Possible, although not very probable, most common soundcards (of the built-in type with a Realtek chip) run natively in 24-bit 96kHz. It actually is easier for systems with these chips to convert a 24-bit 96kHz signal than a 16-bit 44.1 kHz signal. But there are exceptions of audio chips that are limited to 16-bit 44.1 kHz.

The parameter settings I gave are so that the effect processing is set for undoing the audio degeneration of mp3 compression as well as possible. Of course anyone can adjust to taste.

Why 24-bit 96kHz? Because it's the native rate of most DACs in common integrated audio chips, so maximum efficiency with minimum overhead. If you have a premium DAC, you could upsample higher, but it would not bring a lot of benefit, it's still only upsampling of a processed signal. There is no way to undo all the damage that mp3 compression does, in fact, a lot of recordings since the advent of mp3 are quite crappy also in comparison to old studio recordings, in that for most modern music, the total dynamic range of the mastered record is often limited to 6-10 dB, which is highly compressed (for a more energetic sound, it's the same thing Beats Audio does basically). In traditional classical recordings, there is as much as 80 dB of dynamics in a recording, and in modern high quality classical recording, there may even be more dynamic range than that. A recent Metallica album has a dynamic range of about 6 dB. dB is a logarithmic scale, in the normal target range of a recording, 4 dB difference is twice the signal amplitude, so 6 dB of range is a very limited dynamic range for any type of music, and not such a great thing, but hey, it sells...

I have no idea, but I don't think it's the same program as the linux version, because windows can't do LADSPA, so there must be a difference.

You'll never get the same sound quality out of a Windows machine that you would get out of a linux or OSX machine. That has everything to do with the software, and there is no solution for it. Windows just doesn't do audio well. Any studio will tell you that everything sounds better on a Mac or a professional computer console (which is mac or linux), it's not a fairy tale, the difference is huge. A budget solution for better sound without using linux or osx on your PC, is a Raspberry Pi, it's cheaper than even cheap Chinese mp3 players and external soundcards, and you can hook it up to your LAN and control it from your Windows computer, even power it from a USB port, it will even manage your media library and act as a media center, but it will run Raspian or Pidora, and will give a better audio quality than any amount of software or hardware on a windows PC will ever be able to deliver.

well ironically I'm going to be using my notebook as a dedicated Linux machine for media (still have not figured witch distro would be best for the purpose yet though) so i will be hooking up a dac to it any who's as i have a spare pair of mission 220's speakers and amp. I'm not questioning the sound quality difference from windows to Linux/OSX that much im fully aware of (i did music tech at college) still hate logic pro though. was just testing the software out too if there's a noticeable difference and even without the sox and soxr add ons im quite impressed so far.  just wanted to see if i could get em working is all.

I get that with 24-bit 96kHz - i overlooked the native rates of common DACS. But i don't believe that the audacious effects are correcting errors of mp3-compression. - you are just processing your music further more so you like it better. (and maybe compensate some room-acustic-effects and ear-charakteristics and so on and so on) - but there is no main difference to using an EQ (methodicly, technicly there is a great difference, as you said already). But you will effect the music in many ways that were not intended by the producer and the musicians. (and therefor you dont restore a original state (or come near to that))

 

The compression you are talking about (concerning the dynamic-range) is from a different kind. (if i dont miss something) That is a Mixing-Problem and not so much depending on the representation of the music in a file format.

You could turn up your amp and get both...

btw. the large dynamic range in classical music is not at all pointless.

speaking of linux distro's, what would you guy's recommend for media stuff like music, video, picture's etc?

nobody at all? c'mon i thought you guys were bananas for Linux around here?

Distros aren't really software-specific, so whatever distro floats your boat is good for media.

 

You could buy/build a better amplifier. And pushing the RMS-peak to the max doesnt solve your problem.

Classical music is a big field - you are either narrow-minded or you don't know what you are talking about.

well yes, but support varies from distro to distro like drivers, flash and such.different distros use different gui's so im pretty sure some would be inherently better geared towards media comsumption than others after being setup and tweaked of course. i only use linux distro's like parted magic and anti-virus/malware/spyware boot disks for it's utility. so as for as media and general use i would not have a clue to be honest, hence why im asking. after all people have got have there preferences and reasons why if you catch my drift.