Date: Sat, 29 Oct 2005 12:46:50 +0200 From: Hans Petter Selasky <hselasky@c2i.net> To: freebsd-multimedia@freebsd.org Cc: Alexander Leidinger <Alexander@leidinger.net>, skywizard@mybsd.org.my Subject: Re: uaudio and Digigram UAX220 Message-ID: <200510291246.51699.hselasky@c2i.net> In-Reply-To: <20051027133448.vptim01kwoco8wcw@netchild.homeip.net> References: <4358CB6F.2080901@intersonic.se> <20051027.040902.343165902.kazuhito@ph.noda.tus.ac.jp> <20051027133448.vptim01kwoco8wcw@netchild.homeip.net>
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On Thursday 27 October 2005 13:34, Alexander Leidinger wrote: > Kazuhito HONDA <kazuhito@ph.noda.tus.ac.jp> wrote: > > I had a patch which make USB audio device treat 24-bit on FreeBSD. > > It was attached to this mail. > > But cyclic noise will interrupt, > > though FreeBSD sound system can play/record sound source. > > It may result from USB system in FreeBSD, > > and the patch in (http://home.c2i.net/hselasky/isdn4bsd/privat/usb/) > > will resolve this noise problem. > > I don't think we will use the new USB stack if NetBSD/OpenBSD doesn't use > it too. Apart from that, the patch looks nice. If nobody beats me, I try to > commit it at the weekend (no promises, I'm moving and the PC will be moved > at the weekend...). > > Some comments inline. > > > --- uaudio.c.orig Thu Apr 28 02:16:27 2005 > > +++ uaudio.c Tue Sep 6 23:10:09 2005 > > > > @@ -3876,7 +3900,7 @@ uaudio_query_formats(device_t dev, u_int > > } > > } > > > > - if ((pn > 8*2) || (rn > 8*2)) > > + if ((pn > 12*2) || (rn > 12*2)) > > break; > > } > > pfmt[pn] = 0; > > --- uaudio_pcm.c.orig Thu Apr 28 02:16:27 2005 > > +++ uaudio_pcm.c Tue Sep 6 23:10:32 2005 > > @@ -51,13 +51,13 @@ struct ua_info { > > struct ua_chinfo pch, rch; > > }; > > > > -static u_int32_t ua_playfmt[8*2+1]; /* 8 format * (stereo or mono) + > > endptr */ > > +static u_int32_t ua_playfmt[12*2+1]; /* 8 format * (stereo or mono) > > + endptr */ > > Wouldn't it be better to use a named constant instead of the hardcoded > value? We would only have to change one place instead of several in case we > need to change it. Additionally it is more descriptive for the casual > reader. > > Note: you forgot to change the number in the comments... in the recording > part too. > > > -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? I have some comments. The current USB sound driver only uses one isochronous buffer, that is restarted when it is completed. This will lead to a short period of time, +1ms, where no sound data is sent to the external USB device. Depending on the load of your computer, this can be as much as 50ms. So the USB sound driver must use 2 isochronous transfers. At the beginning one will queue both. Then these are restarted on completion. This will result in a constant-rate data stream to the external sound device, a minimum sound buffer equal to the size of the isochronous buffer, and possibly the sound will reach your ears with less delay. Little delay is a result of constant data rate. Currently only my USB driver will support that. If one tries that with the USB driver in *BSD, then it will crash at the first moment one gets a buffer underrun. --HPS
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