First, for those of you who don't know, the SoundBlaster Audigy 2 NX is a usb (2.0) sound device based on the Audigy 2. Most of the features don't appear to be supported--for example EAX and CMSS, and probably not the remote control/IR device.
This device is only supported under ALSA using snd-usb-audio. I am running on Fedora Core 1, and highly recommend reading the Fedora Multimedia Installation Howto at http://www.tldp.org/HOWTO/Fedora-Multimedia-Installation-HOWTO/ and any documentation from the excellent Planet CCRMA site at http://www-ccrma.stanford.edu/planetccrma/software/ which covers several important details. Excerpted from Planet CCRMA is the minimum that I had to modify from just installing the alsa RPMs to get my device to work with any consistency: ----- begin excerpt ---- The USB startup code runs very early in the boot sequence, before the alsasound startup script gets a chance to run. So you have to disable the automatic load of OSS USB modules that will be detected by the hotplug subsystem. To do that add these two lines to the ``/etc/hotplug/blacklist'' file: audio usb-midi This will disable only the USB OSS drivers. You can also add: snd-usb-audio so that your USB soundcard will be started with the rest of the ALSA soundcards (if there are more). Otherwise hotplug will start it early in the boot sequencer (but there's no harm in that). ----- end excerpt ---- Please note that I am not trying to do anything exotic--I don't have much used for Surround, as I do almost all my listening via either a reference-grade headphone amp and Etomotic ER-4S headphones (sweet!) or through an ancient Sansui 4000 "solid state" receiver and a set of modest speakers I built in college (based on 8" klipsch drivers, not bad all told). However, I do have the desire to record radio programs that are always on at terribly inconvenient times. I'll address what I did for that as well. Once the appropriate kernel modules are loaded--the relevant portions of my lsmod are: snd-usb-audio 56832 0 snd-rawmidi 18080 0 [snd-usb-audio] snd-seq-device 5976 0 [snd-seq-oss ...] snd-pcm 84772 0 [snd-pcm-oss ...] snd-page-alloc 9716 0 [snd-pcm] snd-timer 19300 0 [snd-seq snd-pcm] snd 46020 0 [snd-pcm-oss ...] you must run an alsa-enabled mixer to unmute the output. alsamixer worked fine for me, you may prefer the relatively arcane alsactl. So I tried to play wave files and mp3s back through xmms, and everything sounded simply awful. There was a warbling or poping sound, which I typically associate with rate problems. The solution to this problem was pointed to by http://www.qbik.ch/usb/devices/showdev.php?id=2375 which states the Audigy 2 NX "works correctly at 22050, 48000 hz sample rate with or without OSS emulation. Other rates can be up/down sampled using alsa devices." Since everything under the sun is sampled at 44.1 KHz, we need to do enable software resampling. This is accomplished through virtual devices in ALSA. The unbelievably poor documentation indemic in the ALSA programming world (I've seen word that they will be improving soon) doesn't actually say how to do this with clarity. I hacked from one of their example files my ~/.asoundrc to contain ----- begin .asoundrc ----- pcm.48000Hz { type plug # Automatic conversion PCM slave { # Slave definition pcm "hw:0,0" # Slave PCM name rate 48000 # Slave rate (default nearest) or "unchanged" } route_policy default # route policy for automatic ttable generation } ---- end .asoundrc ----- which creates a device called 48000Hz that delivers 48KHz sample signal to (or from?) the sound device. I am able to read from the device without trouble, though arecord works fine without any modification as long as the sample rate is 48000. Then, install the alsa-plugin for xmms. Using the configuration dialog for the plugin set the device name to 48000Hz. Since then I've had no trouble whatsoever with clean playback. I tend to want to record and playback radio programs, so I provide some appropriate signal on the "line in", fiddle with the mixer until I get appropriate levels, and then save the levels with alsactl -f <somefile> store. Recording to mp3 (yeah, ogg would be fine, but the SlimX folks have done nothing with ogg firmware for the 330. Dammit.) is accomplished with a command like arecord -f dat -d $SECONDS 2>>$LOG | /usr/bin/lame -x -r -s 48.0 -q 5 --resample 44.1 --tt "$ID3NAME" - $OUTPUT_DIR/${FNAME}.mp3 >> $LOG 2>&1 The arecord command line dumps little-endian stereo 16 bit PCM data for a duration (-d $SECONDS) to the standard output. The lame line records raw pcm (-r) which in my system required, for some unknown reason, to byteswap (-x) at an input sample rate of 48000 Hz (-s) a quality level of 5 (-q) to prevent pipe overflows, resampled to 44.1 KHz (--resample, probably still grossly overkill for FM stereo broadcasts...thoughts anyone?) gives the file a tag title (-tt) and records the stdin (-) to some output.mp3 file. Fun eh? Features I would really like, anybody know how to do this?: A) enable the remote control and the on-card controls, which consist of a volume knob, a mute, and a CMSS button. If I could get a device to read from (like /dev/irtty or whatever) I would gladly decode the words coming from the respective controls. B) names in the mixer that correspond to the names on the card. My device has names like "mic in", "side/phones", "line in", "csub", "rear", and "front" along with optical and digital I/O. I would love a mapping in alsamixer and GNOME alsa mixer that change "PCM" "PCM" "PCM" "Speaker" "Speaker" "Speaker" into something that means something. C) To be able to use the sound shapers (CMSS and EAX) which for a headphone user are unbelievably sweet! I use the card on a windows box as well, and the EAX and CMSS are really, really good. They put the crap at earwax to shame (try sox on Dark Side of the Moon with earwax, the bells clip and you want to take an ice pick to your ears). If anyone has any interest in helping develop some of the features, please feel free to cross-post to the developer's site. Yours, Park
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