Hi,
I'm trying to build a low-power linux VOIP workstation for replacing the
aging dell workstations here at work. I'm using an EPIA sn18000g board
with compact flash for the storage. I've got Ubuntu 8.10 (minimal
install with xfce4) on it and everything except sound is working great.
The problem I'm having with sound is that when recording at 48000 Hz
(which is the native recording samplerate, I think) I get these odd
echoing pops after I say anything and when I record at 8000 Hz (as
needed for VOIP) the audio quality becomes unbearably staticy.
Additionally, there's a constant background whine, but I can get that
under control by playing with the 'Capture' and 'Digital' input mixer
settings.
It's not a fault of the actual hardware, as FreeBSD on the same hardware
gives crystal clear VOIP (even using the same custom softphone). I'm not
sure if downsampling is just magnifiying the noise I get or if
downsampling itself is causing the problem.
I've tried using the latest ALSA release (1.0.17rc2) candidate with a
new 2.6.25.7 kernel because I saw a lot of intel-hda fixes go in, but
they didn't improve matters. I did get the following warnings in dmesg:
hda-intel: Invalid position buffer, using LPIB read method instead.
hda-intel: IRQ timing workaround is activated for card #0. Suggest a
bigger bdl_pos_adj.
So I added 'position_fix=1 bdl_pos_adj=64' to the module load parameters
and those messages went away but the sound issues remain. I have no idea
what a valid value for bdl_pos_adj is, so I just doubled what it was
defaulting to. I also tried setting 'enable_msi=1', just because it
looked like the card supported it, but it also seems to have had no
effect.
Here's the output of lspci -vvnn:
80:01.0 Audio device [0403]: VIA Technologies, Inc. VIA High Definition Audio Controller [1106:3288]
Subsystem: VIA Technologies, Inc. VIA High Definition Audio Controller [1106:3288]
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR-
Latency: 0, Cache Line Size: 32 bytes
Interrupt: pin A routed to IRQ 17
Region 0: Memory at febfc000 (32-bit, non-prefetchable) [size=16K]
Capabilities: [50] Power Management version 2
Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
Capabilities: [60] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable-
Address: 00000000 Data: 0000
Capabilities: [70] Express Unknown type IRQ 0
Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag-
Device: Latency L0s <64ns, L1 <1us
Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported-
Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop-
Device: MaxPayload 128 bytes, MaxReadReq 128 bytes
Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0
Link: Latency L0s <64ns, L1 <1us
Link: ASPM Disabled CommClk- ExtSynch-
Link: Speed unknown, Width x0
Basically I'm looking for any suggestions here. I can provide any
additional information or perform any testing on request. I will even
provide developer ssh access to the box if it'd help.
My recording testing was done with arecord -vv [-fdat]. The default of
8 kHz sounds terrible, the -fdat format is pretty good except for the
crackling/popping.
Thanks in advance,
Andrew
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