My sound card supports hardware upsampling i.e. I do not need to use the
software up sampling done by plugins like dmix etc. For testing hw up
sampling, I tried to play an 8K sample rate wav file as follows, but some
how it is being upsampled, I think I am missing something.
# aplay -Dhw:0,0 8k.wav -v
Playing WAVE '8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo
Warning: rate is not accurate (requested = 8000Hz, got = 44100Hz)
please, try the plug plugin
Hardware PCM card 0 'OMAP850 EAC' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 2
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 22052
period_size : 5513
period_time : 125011
tick_time : 10000
tstamp_mode : NONE
period_step : 1
sleep_min : 0
avail_min : 5513
xfer_align : 5513
start_threshold : 22052
stop_threshold : 22052
silence_threshold: 0
silence_size : 0
boundary : 1445199872
I have attached my alsa.conf.
-pharaoh.
Attachment:
alsa.conf
Description: Binary data
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