From: "chris hermansen" <clhermansen@xxxxxxxxx>
To: "Wiebe Cazemier" <wiebe@xxxxxxxxxxxx>
Cc: "Clemens Ladisch" <cladisch@xxxxxxxxxxxxxx>, alsa-user@xxxxxxxxxxxxxxxxxxxxx
Sent: Thursday, 16 May, 2013 11:30:08 PM
Subject: Re: Asus Xonar DX (AV200) dmix resampling>But don't they do this because cards don't hardware mix anymore?
> 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).
Hmm, that's a good question. An even better question: how does the player do it when it's talking to the ALSA hardware directly?
Another question that comes to mind, is how do Windows and MacOS do it? I remember from long ago something that Windows 2000 did. I was listening to a song, which sounded fine. As soon as I also opened some low-sample rate wav file, the song that was playing suddenly sounded like 8 kHz as well.
It's just one observation, but perhaps Windows 2000 knew not to resample when there was only one source.
"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.
I can configure my Audacious (my player of choice) to play on hardware as well. However, I will look into Pulse. If what clemens said is true (that it will resample to the frequency of the first song) I can live with it (provided I disable gapless playbak).
I just can't believe Pulse wouldn't be designed with all this in mind.
And why don't soundcard designers just provide actual indepedant channels, that can be mixed in the analog domain, independant of sample rate, like the archaic Amiga sound chips of the 80's and 90's (which was therefore loved my MOD trackers)?
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