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Bit Perfect Audio from Linux - Page 9

post #121 of 294

anyone know if clementine can produce bit perfect playback? i really like the layout, but know very little about whats going on behind the hood with the audio.

post #122 of 294
Quote:
Originally Posted by kchapdaily View Post

anyone know if clementine can produce bit perfect playback? i really like the layout, but know very little about whats going on behind the hood with the audio.

Downloaded Clementine just for ya.

Screenies - 

 

Just specify your sound device in the output device box like this (ex hw:0,1 - don't forget to adjust for your device) for exclusive ALSA playback.

post #123 of 294

Hey guys,

 

I've been following head-fi for quite some time now, but never actually bothered to create an account. But since I see there are a lot of informed fellow *Nix'ers here, I'd like to pop a question that I never could clarify.

If my single goal was to listen to some quality music, would low-latency or even a real time kernel effect my listening experience in any way?

I couldn't quite figure out if low-latency was only relevant to music production or audio in general.

In theory, I wouldn't mind delay if the quality stayed the same.

But then again, you never know...

I'd appreciate any comments, thanks.

post #124 of 294
Quote:
Originally Posted by ruzzaford View Post

Hey guys,

 

I've been following head-fi for quite some time now, but never actually bothered to create an account. But since I see there are a lot of informed fellow *Nix'ers here, I'd like to pop a question that I never could clarify.

If my single goal was to listen to some quality music, would low-latency or even a real time kernel effect my listening experience in any way?

I couldn't quite figure out if low-latency was only relevant to music production or audio in general.

In theory, I wouldn't mind delay if the quality stayed the same.

But then again, you never know...

I'd appreciate any comments, thanks.

 

Real time kernels are only necessary when the computer is in the middle of the audio chain. In other words, it's only necessary if the audio signal comes into the computer trough a sound device, gets mixed, and sent back out trough an audio device. For example, it's necessary if you use a computer as mixer or effect generator during a live event. In this case, the sound from the live instruments goes trough the computer before getting at the speakers - you need the lowest latency possible.

 

If you're using the computer as a source of music, then you really don't need real time kernel. The latency is bellow 50 ms even with a normal kernel, which is acceptable even if the audio is synchronized with a video, which is the only time when timing will be relevant in any way. Timing is irrelevant if you're simply listening to music and will have no influence on sound quality.

post #125 of 294
Wow. That was the perfect answer for me. I searched a lot of forums for this, but never could find a solution because I didn't really know what to Google for. Thanks, Kim!
post #126 of 294
Quote:
Originally Posted by TwinQY View Post

Downloaded Clementine just for ya.

Screenies - 

 

Just specify your sound device in the output device box like this (ex hw:0,1 - don't forget to adjust for your device) for exclusive ALSA playback.

awesome thanks. ill have to double check tomorrow (later today technically), but i was having trouble with some parts of the settings being grayed out.

post #127 of 294
Quote:
Originally Posted by TwinQY View Post

Downloaded Clementine just for ya.

Screenies - 

 

Just specify your sound device in the output device box like this (ex hw:0,1 - don't forget to adjust for your device) for exclusive ALSA playback.

Linux noob here. How do I specify my output device? Like what would I type in the box?

post #128 of 294
Quote:
Originally Posted by anoxy View Post

Linux noob here. How do I specify my output device? Like what would I type in the box?

aplay -l lists devices available. Syntax is as such - X;Y (where X = the card, and Y = the device). 

post #129 of 294
What distro's are you all running? Is there support for the E17 in any or all?

Thanks!

Sent from my HTC Desire HD A9191 using Tapatalk 2
post #130 of 294
Quote:
Originally Posted by TrollDragon View Post

What distro's are you all running? Is there support for the E17 in any or all?

Thanks!

Sent from my HTC Desire HD A9191 using Tapatalk 2

It's the Linux kernel. i.e. in the kernel. For most parts "sound device drivers" are distro-agnostic. Just need to have USB drivers enabled, for example. Again, if you're not compiling your own kernel it matters little.

 

tl;dr - don't worry about incompatibility.

post #131 of 294

If the E17 is USB audio class compliant, then it doesn't even need a driver. At least that's on normal OS that care about standards... That you need driver to use class compliant devices on Windows just goes to prove how ridiculous Microsoft's logic is.

 

If it's not class compliant, then you're kinda left on your own. It's a hit-and-miss thing where you are never guaranteed of a fully working device. In this case, the distro may have an impact on the solution.

post #132 of 294

Hi, all.  I'm new to the audiophile world and to this website.  Sorry if this has been discussed - I can't find it by searching (maybe am using the wrong terminology).  Has anyone had any luck playing 88k/24bit files through a DAC that only supports 96k/24bit (I have the Fiio E17)?  I have tried Gmusicbrowser and Guayadeque - both of which will play the files at 96/24 but they are sped up or at a significantly higher pitch than they should be.  I can't find any way to adjust the pitch in these players (not even sure that would compensate adequately, or if it would degrade the signal).  Any suggestions?

post #133 of 294
You may use deadbeef. It can output 88khz at 96khz. Go to preferences, DSP tab, add, and select Resampler. Then click configure. Enter 96000 as target freq. Aso select best quality/alg.

https://launchpad.net/~starws-box/+archive/deadbeef-player
Edited by eimis - 6/7/13 at 9:18am
post #134 of 294
Quote:
Originally Posted by draiiinage View Post

....playing 88k/24bit files through a DAC that only supports 96k/24bit.....  Any suggestions?

You'll need to convert the sample rate. For a really high quality sample rate conversion you can use sox.

Example command to convert to 88/24:
Code:
sox --no-clobber -G <infile> -b 24 <outfile> rate -v 88200 dither -s

the --no-clobber prevents accidentally overwriting
-G (automatic clip prevention) and dither will together give you the result with the widest possible dynamic range and no clipping.
post #135 of 294
Quote:
Originally Posted by eimis View Post

... deadbeef. It can output 88khz at 96khz. Go to preferences.....Aso select best quality/alg.

Very useful info, thanks.

I haven't compared or tested but it seems that Sox resampler is now considered better than SRC, as used in DeadBeef and many other apps and ALSA add-ons. Anyone have any experience or testing of this?
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