Date: Fri, 28 Oct 2005 22:40:49 +0900 (JST) From: Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp> To: Alexander@Leidinger.net Cc: freebsd-multimedia@freebsd.org, skywizard@MyBSD.org.my Subject: Re: uaudio and Digigram UAX220 Message-ID: <20051028.224049.343190967.kazuhito@ph.noda.tus.ac.jp> In-Reply-To: <20051027133448.vptim01kwoco8wcw@netchild.homeip.net> References: <435B4EBE.5030502@intersonic.se> <20051027.040902.343165902.kazuhito@ph.noda.tus.ac.jp> <20051027133448.vptim01kwoco8wcw@netchild.homeip.net>
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From: Alexander Leidinger <Alexander@Leidinger.net>
Subject: Re: uaudio and Digigram UAX220
Date: Thu, 27 Oct 2005 13:34:48 +0200
> Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp> wrote:
>
> > -static struct pcmchan_caps ua_playcaps = {8000, 48000, ua_playfmt, 0};
> > +static struct pcmchan_caps ua_playcaps = {8000, 96000, ua_playfmt, 0};
>
> You also enhanced the capabilities for playing and recording. I'm not very
> familiar with this (sound system capabilities handling and the USB audio
> standard), but isn't this a property of the underlying hardware and should
> be probed?
The protocol of USB audio doesn't decides sampling rate,
and sampling rates are various according to the devices.
For example, I have three USB audio devices.
One's sampling rate is 48 kHz.
another supports 44.1 kHz and 48 kHz.
The end can be use at 44.1 kHz, 48 kHz and 96 kHz.
So Sampling rate should be probed.
Though I don't know a correct method for sending
the sampling rate to the sound system,
Mathew KAnner has already written the patch for this problem.
http://docs.freebsd.org/cgi/getmsg.cgi?fetch=3877+0+archive/2005/freebsd-multimedia/20050424.freebsd-multimedia
This patch includes solutions for other problems.
Because it is old, it must be changed along recent codes.
Sincerely yours,
Kazuhito HONDA
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