Hi, I recently bought a new motherboard, which, completely to my suprise, had on-board audio that actually was supported, so I could bounce my Soundblaster Live that had been around for years and which I had only be using for it's optical sp/dif output that actually works from linux ;-)
My new setup can generate optical sp/dif by it's own, no need for an extra SBLive card, it sort of works, but I am really puzzled about some things, none of these questions are answered by the docs/faqs, probably due to the codec being very new. Okay, here come te questions, I hope someone can help me out! 1. When the default pcm (hw:0,0) is fed with 44.1 khz audio, the sp/dif link still is at 48 khz. The audio is being resampled somewhere, although I cannot find where. This is both using sox -t alsa and using aplay -D hw:0,0. If I understand correctly, the directive hw:0,0 should bypass any resampling performed by alsa. And the resampling sounds _bad_! It's awful. I suspect it's no more than a linear interpolation. Can it be that it's (badly) incorporated into the ALC882 design itself? 2. Aplay -l lists these devices **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC882 Digital [ALC882 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 How should I interpret the second device? This suggests that the first device is an analogue codec (like the sigmatel on the soundblaster) and the second one (device 2?!) is a straight-pass-through interface. Of course I'd like to use the second one, I don't want alsa or the ALC to do any modification to the bitstream. I can only get useful sound of out device #0 though. When I send any audio data to device #2 (either using hw:0,2 or plughw:0,2) it's messed up. Audio is coming from one channel only, clipped and only bits and pieces are audible. The datasheet seems to show that the ALC882 doesn't have any connection between the analogue part and the sp/dif output. I am really puzzled here. 3. Iecset only allows me to set the sample rate to 32 khz, 44.1 khz and 48 khz. In all other cases, it reverts to 44.1. Anway, it doesn't matter, because the sample rate simply remains at 48 khz and isn't changed. The iecset interface does work though, because if I set the "no audio" flag, the sound is muted. It becomes even more interesting; according to the specs, the ALC882 doesn't even support 32 and 44.1 khz, although it support 96 khz and 192 khz, but iecset (or the underlying logic) won't let me use these. 4. Similarly I'd like to output the complete 24 bits from the mp3 decoder to the (external) dac. Iecset lets me set the word length, but the stream is not affected by it. Actually it all comes down to that I want to have a device that I can feed 96 khz (and preferably other rates as well) audio, 24 bits wide and outputs that unaltered to sp/dif. I know the ALC882 can do it! Thanks.
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