[Solved] Alsa with 24 bit Output Exclusively

Question

This might be an unusual request hoping to find any ALSA expert. Hope this is a right place to post.
I just made an asynchronous USB2 audio device supporting 44.1k, 48k, 88.2, 96k, 192k sampling frequencies. The firmware so far just supports 24bit word length. Since this is a truly audiophile device the mixer / re-sampler is no option although it works (e.g. Amarok with Xine backend). MPlayer plays bit-perfect as long as the content is 24 bit and one of the above sampling frequencies - works since ALSA update in February 2011. It is really fun this way! However, most peoples's music content is 16 bit.
For Windows the ASIO driver is solving the problem, but I doubt it transfers bit-perfect (Tests to follow soon).
I really would like to recommend Linux for the use, but in this case the problem must be resolved. i could imagine 2 ways of doing:
1. Simple: ALSA is configured to output any content with 24 bit word length. Means adding zeros after the 16th bit.
                Disadvantage: Following Digital to Analog converters would accept the signal as true 24 bit and skip the quantization smoothing.
2. Perfect: If there is an available option for doing this re-quantization in Alsa or MPlayer it would be perfect. As long as the algorithm is good, it doesn't matter this process is done within the DAC or within the computer.
My goal is to make a clear instruction of how to use our device with Linux without compromise.
Thanks for any idea!
Last edited by jcci (2011-05-04 02:41:04)

Answer 1

First of all thanks for so much response!
LADSPA was not known to me and I will study what I can do with it. Do you know how to use it?
Edit: LADSPA is meant for production software. No popular Linux player can use those plugins. A solution that works with Mplayer, gstreamer or xine would be accepted by most Linux users, since there are a lot of players on top of those.
Every modern DAC is using re-quantization, synchronous up-sampling and depending on the architecture also noise shaping. The general requirements for s/n can not be achieved with a pure 16bit approach. This is not a problem of the DAC hardware. The USB device can not be initialized when using a 16bit audio file; so the music never reaches the DAC.
The re-sampler inside Linux (or Windows) is definitely downgrading the audio quality. You don't need to have "golden ears" for experiencing this, rather than just a good audio system.
A clean re-quantization could do good, depending on the algorithm.
In case of just enlarging the word length (means in fact no signal change) the re-quantization would be also skipped in the DAC - a very puristic approach which should be proofed to be better. So how to configure ALSA for this?
A dither below on the very low bits of the 24bit signal should be a good idea since it is proofed to statistically shuffle the linearity errors and make the less audible.
Today I did some tests with a very high grade audio system and Linux performs great with 24 bit content. It is not easy to beat top grade CD players on the same DAC. In fact this USB solution is part of a DAC.
The Windows driver works good on my desktop system, but gets choppy on my Macbook... Still some work to do before I can compare.
I admit the cleanest solution would be a firmware revision, but the original firmware making company has no interest in solving this problem and Linux user would have to wait too long or sacrifice sound quality by re-sampling.
Last edited by jcci (2011-04-28 08:42:31)