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I've just bought a new USB DAC (iBasso D4 Mamba, dual Wolfson WM8740 DAC chips) to replace my old USB DAC (Firestone Audio Fubar III, Burr-Brown TI PCM2702). A USB DAC is essentially just an external USB soundcard. With my old DAC, alsamixer presented two sliders; 'master' & 'PCM'; I could control the volume using 'master'. However with my new DAC, alsamixer only presents one slider; 'PCM'. Adjusting 'PCM' does absolutely nothing to the volume. This means that the volume is always at 100%. This is a problem, because the DAC has a recognised flaw, where a lack of voltage to the chips themselves means it clips if it is fed 100%. So I need to drop the volume to ~80% but obviously I can't do this without a functioning volume control!
Another thing that worries me is my /proc/asound/cards entry. I'm not sure why it mentions 'Burr-Brown from TI', because that's not what my new DAC is. I'm worried that drivers/settings from my old DAC have been preserved & I want all traces of it completely purged.
[cj@yuzuki ~]$ cat /proc/asound/cards
0 [default ]: USB-Audio - USB Audio DAC
Burr-Brown from TI USB Audio DAC at usb-0000:00:1a.0-2, full sSome other output that might be useful.
[cj@yuzuki ~]$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: default [USB Audio DAC ], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0[cj@yuzuki ~]$ lsmod|grep '^snd' | column -t
snd_hda_intel 21906 0
snd_seq_dummy 1447 0
snd_hda_codec 76595 1 snd_hda_intel
snd_seq_oss 28928 0
snd_seq_midi_event 5420 1 snd_seq_oss
snd_seq 50530 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_pcm_oss 39096 0
snd_usb_audio 78639 2
snd_pcm 71653 5 snd_hda_intel,snd_hda_codec,snd_pcm_oss,snd_usb_audio
snd_timer 19660 2 snd_seq,snd_pcm
snd_page_alloc 7201 2 snd_hda_intel,snd_pcm
snd_hwdep 6126 2 snd_hda_codec,snd_usb_audio
snd_usb_lib 17362 1 snd_usb_audio
snd_rawmidi 19235 1 snd_usb_lib
snd_seq_device 5241 4 snd_seq_dummy,snd_seq_oss,snd_seq,snd_rawmidi
snd_mixer_oss 16932 1 snd_pcm_oss
snd 57225 17 snd_hda_intel,snd_hda_codec,snd_seq_oss,snd_seq,snd_pcm_oss,snd_usb_audio,snd_pcm,snd_timer,snd_hwdep,snd_usb_lib,snd_rawmidi,snd_seq_device,snd_mixer_oss[cj@yuzuki ~]$ cat /proc/asound/card0/pcm0p/sub0/hw_params
access: RW_INTERLEAVED
format: S16_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
period_size: 5513
buffer_size: 22050
[cj@yuzuki ~]$ cat /proc/asound/card0/pcm0p/sub0/sw_params
tstamp_mode: NONE
period_step: 1
avail_min: 5513
start_threshold: 16537
stop_threshold: 22050
silence_threshold: 0
silence_size: 0
boundary: 6206523236469964800
[cj@yuzuki ~]$ cat /proc/asound/card0/pcm0p/sub0/info
card: 0
device: 0
subdevice: 0
stream: PLAYBACK
id: USB Audio
name: USB Audio
subname: subdevice #0
class: 0
subclass: 0
subdevices_count: 1
subdevices_avail: 0
[cj@yuzuki ~]$ cat /proc/asound/card0/pcm0p/sub0/status
state: RUNNING
owner_pid : 3624
trigger_time: 12467.886130753
tstamp : 13222.343241858
delay : 20066
avail : 2470
avail_max : 21026
-----
hw_ptr : 33284480
appl_ptr : 33304060I've gotten myself mightily confused trying to fix this, so any help would be greatly appreciated!
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You could try renaming /etc/asound.state to /etc/asound.state.backup and then running alsactl store to regenerate it.
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You could try renaming /etc/asound.state to /etc/asound.state.backup and then running alsactl store to regenerate it.
I already tried deleting /etc/asound.state & then restarting alsa, to no success. I just tried explicitly running alsactl store anyway, but again no luck.
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