At Mon, 25 Feb 2008 15:46:28 +0100, William Juul wrote: > > Hello > > I am new to ALSA and trying to write a new driver for a DAC connected > with PCM to an AVR32 on a NGW100 reference card provided by Atmel. > > The sampling rate I am currently using is 11047and the DAC is > providing 4 channels of 24bit. The HW interface is using DMA to copy > data to RAM. > By studying the audio data in hexdump or in Audacity I can verify that > the sound looks good in intervals of about 30mS, then all channels are > garbled for 30mS. This pattern repeat itself throughout the audio > capture. > > I am not confident I have configured all ALSA parameters properly. > How can I go about fixing/debugging this 30mS intverval problem? Maybe the period size has to be aligned to some value? > > Below is the command I am using. > > > Best regards > William Juul > > # arecord -r 11047 -c 4 -f S24_LE -s 1 -A 100 -d 5 --buffer-size 16384 > -F 21333 -v > test.wav Try to change the period size as well. Takashi > Recording WAVE 'stdin' : Signed 24 bit Little Endian, Rate 11047 Hz, Channels 4 > Plug PCM: Hardware PCM card 0 'AVR32 NGW100 external DAC' device 0 subdevice 0 > pcm->setup: 1 > stream : CAPTURE > access : RW_INTERLEAVED > format : S24_LE > subformat : STD > channels : 4 > rate : 11047 > exact rate : 11047 (11047/1) > msbits : 32 > buffer_size : 16384 > period_size : 490 > period_time : 44355 > tick_time : 4000 > tstamp_mode : NONE > period_step : 1 > sleep_min : 0 > avail_min : 490 > xfer_align : 490 > start_threshold : 1 > stop_threshold : 16384 > silence_threshold: 0 > silence_size : 0 > boundary : 1073741824 > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@xxxxxxxxxxxxxxxx > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
