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I'm trying to configure a microphone. I could record with audacity and with
arecord --device=plughw:0,0Tried writing this on ~/.asoundrc
capture.pcm {
type plug
slave.pcm "default microphone"
type hw
card 0
device 0
}now the output for
arecord -vvv --format=dat --device=plughw:0,0 test-mic.wav Recording WAVE 'test-mic.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Plug PCM: Hardware PCM card 0 'HDA Intel MID' device 0 subdevice 0
Its setup is:
stream : CAPTURE
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 1
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 6755399441055744000
appl_ptr : 0
hw_ptr : 0
Max peak (12000 samples): 0x00008000 #################### 100%
Max peak (12000 samples): 0x00008000 #################### 100%
Max peak (12000 samples): 0x0000732c ################## 89%
Max peak (12000 samples): 0x00007302 ################## 89%
Max peak (12000 samples): 0x000065f5 ################ 79%
Max peak (12000 samples): 0x000067b9 ################# 81%
Max peak (12000 samples): 0x00006bbd ################# 84%
Max peak (12000 samples): 0x00006ee3 ################## 86%As you can see in the percentages it's recording correctly, but with
arecord -vvv --format=dat --device=plughw:0,0 test-mic.wav Recording WAVE 'test-mic.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Plug PCM: Linear conversion PCM (S32_LE)
Its setup is:
stream : CAPTURE
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 1
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Slave: Soft volume PCM
Control: Digital Capture Volume
min_dB: -30
max_dB: 30
resolution: 121
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 16384
period_size : 1024
period_time : 21333
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 1
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Slave: Direct Snoop PCM
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 16384
period_size : 1024
period_time : 21333
tstamp_mode : NONE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 1
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
Hardware PCM card 0 'HDA Intel MID' device 0 subdevice 0
Its setup is:
stream : CAPTURE
access : MMAP_INTERLEAVED
format : S32_LE
subformat : STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 32
buffer_size : 16384
period_size : 1024
period_time : 21333
tstamp_mode : ENABLE
tstamp_type : MONOTONIC
period_step : 1
avail_min : 1024
period_event : 0
start_threshold : 1
stop_threshold : 4611686018427387904
silence_threshold: 0
silence_size : 0
boundary : 4611686018427387904
appl_ptr : 0
hw_ptr : 0
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%
Max peak (2048 samples): 0x00000000 # 0%I get this. I think I have to configure correctly which capture device to use by default on the ~/asoundrc
Thanks for reading me and helping.
Last edited by ricardo01l (2022-03-01 21:30:18)
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I'm not sure what you are actually asking? The config is nonsensical, but yes with plughw you can access a device directly. Do you have specific programs you are having issues with? It does sound like you want a duplex stream that can be accessed via "default" regardless, which would be explained in: https://wiki.archlinux.org/title/Advanc … ure_device
In general "default" is a very loose and not clearly defined concept in ALSA and this is highly dependent on how each and every application decides to handle things. If you want more logical consistency consider pulseaudio or pipewire.
Last edited by V1del (2022-02-17 23:19:39)
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Thanks! Installing pulseaudio worked out of the box. An aclaration: the problem was that alsa was recognizing the mic, but it was choosing the incorrect capture device. I think with https://wiki.archlinux.org/title/Advanc … ure_device, you could configure it, but it's easier to use pulse.
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