Wiebe, Clemens, list:
On Thu, May 16, 2013 at 4:23 AM, Wiebe Cazemier <wiebe@xxxxxxxxxxxx> wrote:
----- Original Message -----But don't they do this because cards don't hardware mix anymore?
> From: "Clemens Ladisch" <cladisch@xxxxxxxxxxxxxx>
> To: "Wiebe Cazemier" <wiebe@xxxxxxxxxxxx>
> Cc: alsa-user@xxxxxxxxxxxxxxxxxxxxx
> Sent: Thursday, 16 May, 2013 9:38:10 AM
> Subject: Re: Asus Xonar DX (AV200) dmix resampling
>
> > Now that high-end cards don't do this anymore, it seems to defeat
> > the
> > purpose to then put it in software.
>
> Modern cards don't do this anymore because *all* operating systems
> already do software mixing.
Hmm chicken and egg how fun!
Another thing to think about is gapless playback. How well does this work when a 44.1/16 song is followed by a 96/24 song? The usual functionality is to mix the tail of one with the front of the other; I guess that means downsampling the latter (or a serious application of magic).
"Can't.... use ... new ... technology..." :)
>
> > I understand it's necessary for mixing sources, but doesn't one
> > like
> > to retain full quality when there is one source?
>
> This is a restriction of dmix; using one predetermined sample rate
> avoids communicating between multiple instances. Modern desktops use
> PulseAudio instead.
I guess I have to give Pulse a try. Choosing the sample rate of the first stream is what would suit me. I don't care what music sounds like while I'm skyping, but when I'm just listening to music, I want original quality.
Wiebe please bear in mind that this is just my (perhaps valueless) opinion, but Pulse will cause you pain and agony.
The /etc/pulse/daemon.conf seems to like the idea of a fixed sample rate (looking at my computer, 44.1khz) and to want to resample to that. I have in the past stumbled upon conversations about configuring pulse so it's bit-perfect but for me the Thing That Works is to use a player like guayadeque or quod libet or something + mpd that will talk directly to Alsa.
While I agree that I find the whole HD audio stuff a bunch of marketing hype (CD quality is good enough for human hearing), I think it too bold of the subsystem to just downmix to 48 kHz, mostly because any resampling theoretically introduces aliasing artifacts.
>
> Anyway, in what way does resampling from 192 to 48 kHz reduce
> quality?
> Are you a bat? ;-)
Being somewhat nervous of accusations of trying to start a discussion worthy only of an audio equipment enthusiast group (at best)....
I am not a bat either (in fact my ears aren't very young) but if I have the choice between some super-processed CD pressing perhaps from a 5th generation analogue copy vs. someone's effort to extract an analogue master at 96/24, or something that was originally recorded at higher resolution, I prefer to avoid the (possible) compression and gain limiting conversion to 16 bits and/or downsample, as well as the (possibly suspect) downsampling on my laptop.
It also seems apparent that DACs that take a little extra care in the analogue section tend to support 96/24 or higher anyway.
For what it's worth!
--
Chris Hermansen · clhermansen "at" gmail "dot" com
C'est ma façon de parler.
C'est ma façon de parler.
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