Hi, My machine has two sound cards, both of which I need to use with Alsa. I ran alsaconf choosing each card individually and writing a modules.conf for each. I have then attempted to merge the two of them together into a single file. The merged file is my current modules.conf, and when booting I see no errors, and see some evidence that code has gotten loaded for each card.
Sep 8 15:01:08 RH74H alsasound: Starting sound driver snd-ice1712 succeeded Sep 8 15:01:09 RH74H kernel: Hammerfall memory allocator: buffers allocated for 1 cards Sep 8 15:01:09 RH74H alsasound: Starting sound driver snd-rme9652 succeeded Sep 8 15:01:09 RH74H alsasound: Starting sequencer driver snd-seq-midi succeeded Sep 8 15:01:09 RH74H alsactl: No state is present for card card1 Sep 8 15:01:09 RH74H alsasound: Restoring sound driver settings succeeded The AP2496 is working fine. alsaplayer plays CDs. My problem at this point is that I am not clear how to send audio out of the RME Hammerfall to an external unit. I'd like to send alsaplayer data first, and then jack/ardour data, via the Hammerfall to other audio equipment. I attach what appears to be relevant info from /proc/asound/card1. It looks like all the pcm channels are off. Do I need to run alsamixer for this device to turn them on? Seems I would. I tried: alsamixer -c card1 and alsamixer -c 1 with no luck. Something about no mixer elems found. So, at this point I'd love someone who knows this stuff more deeply than I to take a look at my modules.conf and see if it looks OK. I do not see any errors in dmesg or /var/log/messages, but I do not know where else to look. Thanks! Mark [root@RH74H card1]# pwd /proc/asound/card1 [root@RH74H card1]# ls id pcm0c pcm0p rme9652 [root@RH74H card1]# more rme9652 RME Digi9636 (Rev 1.5) (Card #2) Buffers: capture eea00000 playback ee800000 IRQ: 10 Registers bus: 0xf5000000 VM: 0xf08e7000 Control register: 4404e Latency: 8192 samples (2 periods of 32768 bytes) Hardware pointer (frames): 0 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 ADAT1 Input source: ADAT1 optical IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz ADAT1: Sync ADAT2: No Lock ADAT3: No Lock Timecode signal: no Punch Status: 1: off 2: off 3: off 4: off 5: off 6: off 7: off 8: off 9: off 10: off 11: off 12: off 13: off 14: off 15: off 16: off 17: off 18: off ************** modules.conf ************** alias parport_lowlevel parport_pc alias eth0 3c59x alias usb-controller usb-uhci alias char-major-116 snd options snd snd_major=116 snd_cards_limit=2 snd_device_mode=0666 alias snd-card-0 snd-ice1712 alias snd-card-1 snd-rme9652 options snd-ice1712 snd_index=0 options snd-rme9652 snd_index=1 alias char-major-14 soundcore alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias sound-slot-1 snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-1 snd-seq-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-8 snd-seq-oss alias sound-service-1-12 snd-pcm-oss # --- Keep modules from being autocleaned add options -k snd-card-0 alias sound-slot-2 snd-ice1712 post-install sound-slot-2 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || : pre-remove sound-slot-2 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || : ------------------------------------------------------- This sf.net email is sponsored by: OSDN - Tired of that same old cell phone? Get a new here for FREE! https://www.inphonic.com/r.asp?r=sourceforge1&refcode1=vs3390 _______________________________________________ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
