This works:
root:/> arecord -Dhw:0 -d 10 -t wav -r 48000 -f S16_LE -c 2 test.wav
This doesn't:
root:/> arecord -Dhw:0 -d 10 -t wav -r 44100 -f S16_LE -c 2 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
asoc: can't set codec ssm2602-hifi hw params
arecord: set_params:1123: Unable to install hw params:
ACCESS: RW_INTERLEAVED
FORMAT: S16_LE
SUBFORMAT: STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: 44100
PERIOD_TIME: (125011 125012)
PERIOD_SIZE: 5513
PERIOD_BYTES: 22052
PERIODS: 4
BUFFER_TIME: (500045 500046)
BUFFER_SIZE: 22052
BUFFER_BYTES: 88208
TICK_TIME: 0
I am not sure exactly what is wrong, I do the same thing on a AD1890
codec and I get this:
root:~> arecord -Dhw:0 -d 10 -t wav -r 41000 -f S16_LE -c 2 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 41000 Hz, Stereo
Warning: rate is not accurate (requested = 41000Hz, got = 48000Hz)
please, try the plug plugin
but it still continues to record. This might not even be a problem,
but I am trying to debug why portaudio works good on the AD1980 codec
and blows chunks on the SSM2603 with the same portaudio lib (slightly
different ALSA lib, the SSM2603 is 1.0.24 and the AD1980 is 1.0.18)
and same app code.
There is a different ALSA config for each codec but I'm having to
learn how the config reads. I could use some help here.
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