Hi Peter, Appreciate your response. Yes, I'm able to hear the mic through the speakers if I unmute the Audigy Analog/Digital Output Jack and turn up the volume on the Mic channel. I didn't hear anything at first, but removed and reinserted the mic cable and it works.
I'm able to record if I use the TAB key in AlsaMixer to see the [Capture] channels and turn up the mix capture volume. Let me add I find it particularly pathetic that I've used Linux for several years now, and spent a lot of time tweaking ALSA, and never realized or noticed that AlsaMixer has a separate display for capture channels. In alsamixergui, there are two mic channels, but afaics they're not labeled in any way. Anyway, this is working well, thank you for a great driver. Best, Dave Peter Zubaj wrote: > Hi, > > Can you hear mic through speakers (headphone) ? > Could you try to record from device 1 (this is not very usable - but it > bypass mixer) ? > > Peter Zubaj > > On Wed, 2006-06-14 at 02:34 -0700, David Liontooth wrote: > >> On an Audigy 2 Value SB0400 I'm unable to record from the microphone. >> >> Sound output is working fine. I plugged the microphone into the red jack >> and tried to record using audacity, >> but get no response. >> >> # arecord -l >> **** List of CAPTURE Hardware Devices **** >> card 0: Audigy2 [Audigy 2 Value [SB0400]], device 0: emu10k1 [ADC >> Capture/Standard PCM Playback] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> card 0: Audigy2 [Audigy 2 Value [SB0400]], device 1: emu10k1 mic [Mic >> Capture] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> card 0: Audigy2 [Audigy 2 Value [SB0400]], device 2: emu10k1 efx >> [Multichannel Capture/PT Playback] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> # lsmod | grep snd >> snd_pcm_oss 46368 0 >> snd_mixer_oss 14848 5 snd_pcm_oss >> snd_emu10k1 110084 4 >> snd_rawmidi 21344 1 snd_emu10k1 >> snd_ac97_codec 98492 1 snd_emu10k1 >> snd_ac97_bus 2560 1 snd_ac97_codec >> snd_pcm 78600 3 snd_pcm_oss,snd_emu10k1,snd_ac97_codec >> snd_seq_device 7440 2 snd_emu10k1,snd_rawmidi >> snd_timer 20360 2 snd_emu10k1,snd_pcm >> snd_page_alloc 8656 2 snd_emu10k1,snd_pcm >> snd_util_mem 4032 1 snd_emu10k1 >> snd_hwdep 8264 1 snd_emu10k1 >> snd 49056 9 >> snd_pcm_oss,snd_mixer_oss,snd_emu10k1,snd_rawmidi,snd_ac97_codec,snd_pcm,snd_seq_device,snd_timer,snd_hwdep >> >> # amixer controls | grep Capture >> numid=8,iface=MIXER,name='PCM Capture Volume' >> numid=9,iface=MIXER,name='Synth Capture Volume' >> numid=17,iface=MIXER,name='Line2 Capture Volume' >> numid=11,iface=MIXER,name='Mic Capture Volume' >> numid=21,iface=MIXER,name='Aux2 Capture Volume' >> numid=15,iface=MIXER,name='IEC958 Optical Capture Volume' >> numid=19,iface=MIXER,name='Analog Mix Capture Volume' >> numid=13,iface=MIXER,name='Audigy CD Capture Volume' >> numid=32,iface=PCM,name='Captured FX8010 Outputs',device=2 >> >> # amixer -c 0 cset numid=11 50%,50% unmute cap >> numid=11,iface=MIXER,name='Mic Capture Volume' >> ; type=INTEGER,access=rw---,values=2,min=0,max=100,step=0 >> : values=50,50 >> >> # amixer -c 0 cget numid=19 >> numid=19,iface=MIXER,name='Analog Mix Capture Volume' >> ; type=INTEGER,access=rw---,values=2,min=0,max=100,step=0 >> : values=50,50 >> >> I run this while talking into the microphone: >> >> # arecord -d 5 -N -vvv foobar.wav >> Recording WAVE 'foobar.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono >> Plug PCM: Linear conversion PCM (S16_LE) >> Its setup is: >> stream : CAPTURE >> access : RW_INTERLEAVED >> format : U8 >> subformat : STD >> channels : 1 >> rate : 8000 >> exact rate : 8000 (8000/1) >> msbits : 8 >> buffer_size : 2048 >> period_size : 1024 >> period_time : 128000 >> tick_time : 1000 >> tstamp_mode : NONE >> period_step : 1 >> sleep_min : 0 >> avail_min : 1024 >> xfer_align : 1024 >> start_threshold : 1 >> stop_threshold : 2048 >> silence_threshold: 0 >> silence_size : 0 >> boundary : 4611686018427387904 >> Slave: Hardware PCM card 0 'Audigy 2 Value [SB0400]' device 0 subdevice 0 >> Its setup is: >> stream : CAPTURE >> access : MMAP_INTERLEAVED >> format : S16_LE >> subformat : STD >> channels : 1 >> rate : 8000 >> exact rate : 8000 (8000/1) >> msbits : 16 >> buffer_size : 2048 >> period_size : 1024 >> period_time : 128000 >> tick_time : 1000 >> tstamp_mode : NONE >> period_step : 1 >> sleep_min : 0 >> avail_min : 1024 >> xfer_align : 1024 >> start_threshold : 1 >> stop_threshold : 2048 >> silence_threshold: 0 >> silence_size : 0 >> boundary : 4611686018427387904 >> Max peak (1024 samples): 0x00000001 # 0% >> Max peak (1024 samples): 0x00000001 # 0% >> >> No sound input appears to be detected. When I play the file, I get a >> low-level static: >> >> # aplay foobar.wav >> Playing WAVE 'foobar.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono >> >> Suggestions? Am I looking at the wrong channels? >> >> David >> >> > > > _______________________________________________ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
