Hi :)
I'm trying to create an .asoundrc file for my box which will combine a
USB sound card output and the SPDIF output on an onboard Intel device.
Thing is, the USB card is a NAD DAC 1 device which _only_ support a
sample rate of 48 kHz. This card provides audio for my bedroom, while
in my livingroom I'm using an AudioAlchemy II DAC which supports both
44.1 kHz and 48 kHz sample rates. The audio going in to my box is
16/44.1 via Airplay, originating from CD-format music, i.e. the same.
Since the livingroom is where my "real" hifi equipment is, I'd rather
not do sample rate conversion on the source if at all avoidable.
Hence, I'm trying to create a config with 48 kHz conversion for the
NAD and none for the AA2.
Problem is, when I try to use it with the SPDIF out at 44.1, I get
"ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not
usable". Works at 48 kHz.
Is this a limitation or am I just doing it wrong? Hope I'm not being a
retard (more than usual, anyway).
I've added my conf below. Hope to hear from someone. Thank you :)
Daniel
~/.asoundrc:
defaults.pcm.rate_converter "samplerate_best"
pcm.hifi plug:both
ctl.hifi {
type hw
card Intel
}
pcm.both {
type route
slave.pcm {
type multi
slaves.b.pcm aa2
slaves.a.pcm nad
slaves.a.channels 2
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
}
ttable.0.0 1;
ttable.1.1 1;
ttable.0.2 1;
ttable.1.3 1;
}
ctl.both {
type hw
card Headset
}
pcm.nad {
I'm trying to create an .asoundrc file for my box which will combine a
USB sound card output and the SPDIF output on an onboard Intel device.
Thing is, the USB card is a NAD DAC 1 device which _only_ support a
sample rate of 48 kHz. This card provides audio for my bedroom, while
in my livingroom I'm using an AudioAlchemy II DAC which supports both
44.1 kHz and 48 kHz sample rates. The audio going in to my box is
16/44.1 via Airplay, originating from CD-format music, i.e. the same.
Since the livingroom is where my "real" hifi equipment is, I'd rather
not do sample rate conversion on the source if at all avoidable.
Hence, I'm trying to create a config with 48 kHz conversion for the
NAD and none for the AA2.
Problem is, when I try to use it with the SPDIF out at 44.1, I get
"ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not
usable". Works at 48 kHz.
Is this a limitation or am I just doing it wrong? Hope I'm not being a
retard (more than usual, anyway).
I've added my conf below. Hope to hear from someone. Thank you :)
Daniel
~/.asoundrc:
defaults.pcm.rate_converter "samplerate_best"
pcm.hifi plug:both
ctl.hifi {
type hw
card Intel
}
pcm.both {
type route
slave.pcm {
type multi
slaves.b.pcm aa2
slaves.a.pcm nad
slaves.a.channels 2
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
}
ttable.0.0 1;
ttable.1.1 1;
ttable.0.2 1;
ttable.1.3 1;
}
ctl.both {
type hw
card Headset
}
pcm.aa2 {
type dmix
ipc_key 1024
slave {
pcm "hw:0,1"
period_time 0
period_size 1024
buffer_time 0
buffer_size 16384
format S16_LE
rate 48000
}
bindings {
0 0
1 1
}
}
type dmix
ipc_key 1025
slave {
pcm "hw:2,0"
period_time 0
period_size 1024
buffer_time 0
buffer_size 16384
format S16_LE
rate 48000
}
bindings {
0 0
1 1
}}
ctl.aa2 {
type hw
card Intel
}
ctl.nad {
type hw
card Headset
}
ctl.aa2 {
type hw
card Intel
}
ctl.nad {
type hw
card Headset
}
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