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@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel