From mboxrd@z Thu Jan 1 00:00:00 1970 Return-Path: Received: from gate.crashing.org (gate.crashing.org [63.228.1.57]) (using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits)) (Client did not present a certificate) by ozlabs.org (Postfix) with ESMTP id A22A6679E0 for ; Sun, 26 Mar 2006 09:07:58 +1100 (EST) Subject: snd-aoa & rates From: Benjamin Herrenschmidt To: Johannes Berg Content-Type: text/plain Date: Sun, 26 Mar 2006 09:06:36 +1100 Message-Id: <1143324396.6318.8.camel@localhost.localdomain> Mime-Version: 1.0 Cc: linuxppc-dev list List-Id: Linux on PowerPC Developers Mail List List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Hi Johannes Current snd-aoa blew up on me at module load btw ... anyway, that's not my point here :) I wonder if you may be doing something a bit too complicated and maybe not perfectly wanted for your bitrates thingy... you basically end up with min/max limits and bit size limits, sort-of assuming that on analog, any rates you have in your list that fits those limits will be available on a given codec. That may not be the case for all codecs no ? Then you end up with a similar list for digital but since you need discrete bitrates, you end up with a list with min==max. Why not do something much simpler which is to define the known bitrates as a bitmask and have the codec expose a bitmask of supported analog and digital rates ? In fact, I would have been even nastier and only exposed the intersection of the above so I don't have to bother about rates that digital won't support :) But I suppose that if you really want to support 8k or 96k it might make sense to support others. Also, for the sample sizes, same comment. Number of bits are not that useful. I'd rather have a bitmask of formats: 8 bits, 16 bits msb, 24 bits msb, maybe lsb versions if supported, ac3, floating point if supported, etc... That or an array. I'm sure Alsa already have constants defined for those no ? I would then have the codec have a function returning the required clocks for a given bitrate/format combination... That is all suggestions of course, if you feel that what you do is better, then stick to it :) Another thing I wouldn't have bothered with is again with whatever digital supports or doesn't ... rather that trying to prevent some rates from being useable by alsa based on a control that users will typically not have means to set at the right time (what about a sound server running all the time keeping the drier running, you want to block the digital switch ?) what I would do is just "mute" the digital output if a format is selected that isn't supported for digital. I would let the user chose the formats they want at all time, and only clamp the digital enable/disable switch. On this switch, btw, you should then remember the user setting: if the user switches it off, remember off. If the user switches it on, remember on, If the user sets it on but you have to mute it, remember that so that when the sample size/format changes again, unmute. Sames goes for things that may be supported by the digital output and not analog (ac3 ?). In this case, mute the analog outputs. The mutes of these are controlled externally via the amps so it may be a bit complicated, unless you define specific messages to the core for that, or maybe just clamp the master volume down in the codec driver. Ben.