Can't capture audio without introducing noise (sound-card line-in or capture card)
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Can't capture audio without introducing noise (sound-card line-in or capture card)
Slackware64 14 and Gentoo.
No matter what I try it seems I can't get a clean recording of audio from soundcard's line-in or capture card; everything works fine in Windows however. Any attempt to record introduces pops, clicks and static that resembles a cheap LP record player. Playback sounds perfect with soundcard line-in, but streaming capture-card audio sounds ridiculously bad (full of screetching, static and sqeaking).
I think the I/O is messed up but I don't know where. My entire system slows down when recording any video stream from a game emulator. When streaming from capture card dmesg gets flooded with buffer overflow messages (video captures result in dropped frames and corruption).
I've tried arecord and audacity to capture audio from sound-card line-in with no luck. The best I've managed is making a buffer of 10 milliseconds (or more) in audacity which gets rid of static and pops, but then the audio skips as it's being captured.
This is really bothersome since I keep playing my laserdisc player repeatedly and wearing it out, without having a successful capture.
Is there a known problem with my AMD AM3+ motherboard (GA-970A-UD3)? I'm using 16gb of ram, motherboard sound, and PVR-150 capture card with ivtv drivers.
Any help would be much appreciated.
Kind regards
EDIT:
Seems like others have reported problems with IOMMU on this GA-970A-UD3 board. Been using iommu=pt which pretty much disables iommu (even if it's enabled in the bios which I do). Could this be related to the problems I'm having?
Iommu is to do with passing pci interfaces to virtual machines. That will not affect your soundcard recording quality.
If you are trying to use just the standard onboard stuff it is expected. The cheap chips used as dac really don't work that well. If possible get dedicated hardware that support ASIO. Then you can have a lot more freedom to change many things. Your initial tuning the buffer is the correct way of fault finding. But also consider bit rate and a few other things.
I have a producer usb mic from maudio. Using it's ASIO basically gets rid of my pops and clicks. Sometimes external noise is still captured, but those can be taken out with filters.
Still have problems with noise. Don't seem to have any problems with iommu or irq conflicts either (loopback sounds great and capture card problems were due to mplayer). Noise is only introduced after arecord, and I noticed it happens more so if period size is set too close to buffer size. Still don't know how to totally eliminate it.
I found audacity records fine without noise with buffer time of 50ms, but that program is too heavy and I can't use it through a script.
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