Weibe, Clemens, list: On Thu, May 16, 2013 at 5:51 PM, Wiebe Cazemier <wiebe@xxxxxxxxxxxx> wrote: > 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 > > >> > >> >> > Modern cards don't do this anymore because *all* operating systems >> > already do software mixing. >> >> But don't they do this because cards don't hardware mix anymore? > > > 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? My point is, I don't believe the player CAN do it. A thought experiment, though you can certainly try it out for real: Enable gapless playback. Cue up a 44.1/16 song. Start it playing. Now cue up a 96/24 song. What is going to happen when the player gets to the point where it needs to blend the last 5 seconds of the first with the first 5 of the second? It's going to either crap out or resample the second song, that's what it's going to do. By definition gapless playback is mixing two data streams together. And there is no way for the player to know at the beginning of a session that three hours later the person using it is going to choose a song with a different bit depth. > > 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. It's my impression that iTunes or what's under it runs at a (pre-configured) bit rate/depth. If you want to play a 96/24 song without resampling when it is already playing 44.1/16 without resampling, you apparently have to stop the player, reconfigure something, and start the player up again. That is why people buy things like Decibel and Amarra. (struggling to avoid editorial comments about all this). > > >> >> 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. Well. I don't know what to say about that. "Belief is good"? "If wishes were horses"? Anyway, I hope that works for you!!! :-) Anyway anyway, remember that Pulse (commonly) sits atop Alsa (or I guess OSS) anyway. So what are you buying by putting Pulse on top? Not sure. > > > 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)? Not sure I know the answer to this, but I would guess that most sound card designers tend to look at all problems with a digital hammer and leave aside as many analogue components as possible. -- Chris Hermansen · clhermansen "at" gmail "dot" com C'est ma façon de parler. ------------------------------------------------------------------------------ AlienVault Unified Security Management (USM) platform delivers complete security visibility with the essential security capabilities. Easily and efficiently configure, manage, and operate all of your security controls from a single console and one unified framework. Download a free trial. http://p.sf.net/sfu/alienvault_d2d _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
