[SOLVED] USB mic works perfectly, line one doesn't

ranko_6 · 10-29-2010, 09:02 AM

When I started using Slackware I've been using USB headset. Everything ran perfectly fine. They got recognized, I could use all of it's functions and so on. Unfortunately volume control on them stopped working so I had to service them, and like in every third world country, yay I need to wait minimum three weeks just so they can press a button and say "Yup, they don't work".
I bought normal headset to use them while other one is getting fixed, and didn't expect any problems with them considering they are "old" technology, normal audio input and output.

Now problem is following, sound works fine but microphone doesn't. Once I managed to "fix" it by switching audio input from mic to something else and then back to mic, now even that doesn't work. Another... "problem" is if I leave my USB webcam plugged in while booting computer, there is no sound at all, ether on line ones or USB ones but I can easily solve that problem by unplugging webcam, rebooting and plugging it back in later.
Now everything ran fine until my computer shutdown while I was sleeping. Don't know why, but it started happening recently, don't see anything smart in logs so I assume its... something, read it could be cache, HDD, temperature and so on but since logs don't write down anything smart... will deal with it some other time.

EDIT
When I talk I can actually hear myself on the headset, which probably means that they work. If I put mic boost all the way up, I can hear myself perfectly clear, but somehow... seems that that information just doesn't get to Slackware, well it gets somewhat considering that it reacts to me adjusting mic boost on KMix, but everything else... no.

Arecord doesn't record anything, TS register as microphone isn't plugged in at all, skype doesn't get any input ether.
EDIT

I've googled little bit about the problem, and although not sure how to deal with it I found which information might be useful so someone might be able to help me.

Code:

aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

Find weird that it counts Analog input twice... maybe that's the problem?

Code:

arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: Intel [HDA Intel], device 2: ALC883 Analog [ALC883 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: U0x46d0x807 [USB Device 0x46d:0x807], device 0: USB Audio [USB Audio]
  Subdevices: 0/1
  Subdevice #0: subdevice #0

I have only one microphone plugged in, other one is webcam. So... why 2 captures? Can I just remove that line from pcm file to "fix" it?

Code:

cat /proc/asound/pcm
00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 1
00-01: ALC883 Digital : ALC883 Digital : playback 1 : capture 1
00-02: ALC883 Analog : ALC883 Analog : capture 1
01-00: USB Audio : USB Audio : capture 1

Code:

lspci -vvnn
00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02)
        Subsystem: Giga-byte Technology Device [1458:a002]
        Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+
        Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
        Latency: 0, Cache Line Size: 32 bytes
        Interrupt: pin A routed to IRQ 45
        Region 0: Memory at f9100000 (64-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 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME-
        Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+
                Address: 00000000fee0300c  Data: 4191
        Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00
                DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us
                        ExtTag- RBE- FLReset-
                DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported-
                        RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+
                        MaxPayload 128 bytes, MaxReadReq 128 bytes
                DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend-
                LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us
                        ClockPM- Surprise- LLActRep- BwNot-
                LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk-
                        ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt-
                LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt-
        Capabilities: [100] Virtual Channel <?>
        Capabilities: [130] Root Complex Link <?>
        Kernel driver in use: HDA Intel
        Kernel modules: snd-hda-intel

Found somewhere that the problem might be solved by running alsaconf so I did, though didn't help

Code:

cat /etc/modprobe.d/sound.conf 
# --- BEGIN: Generated by ALSACONF, do not edit. ---
# --- ALSACONF version 1.0.23 ---
alias snd-card-0 snd-hda-intel
alias sound-slot-0 snd-hda-intel
# --- END: Generated by ALSACONF, do not edit. ---

Code:

dmesg | grep HDA
[    3.996018] HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22
[    3.998432] HDA Intel 0000:00:1b.0: irq 45 for MSI/MSI-X
[    3.998455] HDA Intel 0000:00:1b.0: setting latency timer to 64

Code:

alsactl init
Unknown hardware: "HDA-Intel" "Realtek ALC883" "HDA:10ec0883,1458e601,00100002" "0x1458" "0xa002"
Hardware is initialized using a guess method

That's it, hope someone will make something out of this and thank for your help

T3slider · 10-30-2010, 03:49 PM

I have a similar situation (though I have no USB mic). My microphone is detected as being at hw:0,2 instead of hw:0,0. You should be able to record with arecord using

Code:

$ arecord -f cd -D 'hw:0,2' test.wav

You can also create a ~/.asoundrc file to create an alias that other applications should be able to pick up on. For example:

Code:

pcm.dsnooped {
	type dsnoop
	ipc_key 234884
	slave {
		pcm "hw:0,2"
		channels 2
	}
}
ctl.dsnooped {
	type dsnoop
	ipc_key 234884
	slave {
		pcm "hw:0,2"
		channels 2
	}
}

This dsnooped device works (for me) in arecord and audacity (though audacity let me choose hw:0,2 explicitly anyway). I don't have Skype to test, so I'm not sure if it works there. I am having some difficulty with TeamSpeak 3 though (and I assume that's what you were talking about when you mentioned TS). It doesn't offer hw:0,2 as a choice for me, and entering it manually doesn't help. I can select the default device, but that refers to hw:0,0 which obviously won't work. It *does* present me with a dsnooped option (my alias), but it refuses to work. The TeamSpeak developers have never been very good with proper ALSA support, and so far the only solution I have is *very* hacky. I'm all ears for a better solution, but my searching has done nothing, and any alias or plugin I use in .asoundrc refuses to work in TS despite working in every other application I tried. Anyway, my (TERRIBLE, TERRIBLE) solution [avoid this if you can]:

Before launching TeamSpeak (or Skype, if you can't get it working properly there either, or any other non-conforming application), as root you will need to hack up /dev/snd/*. On my system, I have /dev/snd/pcmC0D0c (a capture device at hw:0,0 that doesn't work) and /dev/snd/pcmC0D2c (a capture device at hw:0,2 that does work).

Code:

# cd /dev/snd
# rm pcmC0D0c
# ln -s pcmC0D2c pcmC0D0c

You won't be permanently destroying anything here since these files are generated each time the snd-hda-intel module is loaded, but this is obviously NOT ideal. I cannot figure out how to rename devices (cards are easy, but *devices*...). I think the reason this isn't possible (or at least isn't documented anywhere I can see) is because it should never be necessary -- any good application should simply let you choose which device to use. Unfortunately TeamSpeak does not.

Hopefully someone will chime in with a real solution, but this hack should at least keep you going until (if ever) that time comes.

It may be possible to just use "pcm.!default" to force the system to use hw:0,2 as a capture device, but my *playback* device is at hw:0,0 and my .asoundrc hacking knowledge is not good enough to create a default capture device that differs from the default playback device. If you can somehow manage that, it *may* work in TeamSpeak 3 (and other weird applications) without needing the above hack -- but I cannot make any promises.