[alsa-devel] new driver: 30mS interval problem
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?
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 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
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
On Mon, Feb 25, 2008 at 4:43 PM, Takashi Iwai tiwai@suse.de wrote:
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
I tried changing the period size in both directions but it would never change to anything but 490. Then I tried changing the number of channels. -c 1 gave me lots of "overrun", but -c 2 made everything work. The WAV file would still contain data from all 4 channels. I do not understand the correlation between the different parameters and why it behaves like this. Thus I do not know what to adjust to remedy this.
The command I used to make everything work was: arecord -r 11047 -c 2 -f S24_LE -s 100 -A 10000 -d 5 > test.wav
It turned out that --buffer-size and -F did not actually change anything.
The problem that is still bothering me is that the different channels from the PCM stream comes in different positions in the WAV file. For example when I disable all but the first PCM channel, the data from this channel can occur in any of the 4 channels in the WAV file (there is still four channels with data in the WAV file but the WAV header says it contains 2 channels when captured with -c 2).
How can I make sure that the data from one particular channel always occur in a fixed position in the WAV file?
Best regards William
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
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
At Tue, 26 Feb 2008 13:56:51 +0100, William Juul wrote:
On Mon, Feb 25, 2008 at 4:43 PM, Takashi Iwai tiwai@suse.de wrote:
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
I tried changing the period size in both directions but it would never change to anything but 490.
Then your driver has some constraints. Or does it use dmix?
Then I tried changing the number of channels. -c 1 gave me lots of "overrun", but -c 2 made everything work. The WAV file would still contain data from all 4 channels. I do not understand the correlation between the different parameters and why it behaves like this. Thus I do not know what to adjust to remedy this.
The command I used to make everything work was: arecord -r 11047 -c 2 -f S24_LE -s 100 -A 10000 -d 5 > test.wav
It turned out that --buffer-size and -F did not actually change anything.
Check with -v option what plugins are being used.
Takashi
On Tue, Feb 26, 2008 at 2:22 PM, Takashi Iwai tiwai@suse.de wrote:
At Tue, 26 Feb 2008 13:56:51 +0100,
William Juul wrote:
On Mon, Feb 25, 2008 at 4:43 PM, Takashi Iwai tiwai@suse.de wrote:
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
I tried changing the period size in both directions but it would never change to anything but 490.
Then your driver has some constraints. Or does it use dmix?
Then I tried changing the number of channels. -c 1 gave me lots of "overrun", but -c 2 made everything work. The WAV file would still contain data from all 4 channels. I do not understand the correlation between the different parameters and why it behaves like this. Thus I do not know what to adjust to remedy this.
The command I used to make everything work was: arecord -r 11047 -c 2 -f S24_LE -s 100 -A 10000 -d 5 > test.wav
It turned out that --buffer-size and -F did not actually change anything.
Check with -v option what plugins are being used.
This is the output with -v. The configuration is printed twice since I have compiled a version of arecord that always print it. I do not use dmix, and I have not intentionally made any constraints. The driver source code is attached.
best regards William
# arecord -r 22786 -c 2 -f S24_LE -s 100 -A 10000 -d 5 -v > temp.wav ALSA /home/williaj/src/ads127x/src/ads127x.c:158: snd_ads127x_pcm_open Recording WAVE 'stdin' : Signed 24 bit Little Endian, Rate 22786 Hz, Stereo pcm->setup: 1 stream : CAPTURE access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 2 rate : 22786 exact rate : 22786 (22786/1) msbits : 32 buffer_size : 4044 period_size : 1011 period_time : 44369 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1011 xfer_align : 1011 start_threshold : 1 stop_threshold : 4044 silence_threshold: 0 silence_size : 0 boundary : 2120220672 Plug PCM: Hardware PCM card 0 'AVR32 NGW100 external DAC' device 0 subdevice 0 Its setup is: pcm->setup: 1 stream : CAPTURE access : RW_INTERLEAVED format : S24_LE subformat : STD channels : 2 rate : 22786 exact rate : 22786 (22786/1) msbits : 32 buffer_size : 4044 period_size : 1011 period_time : 44369 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 100 avail_min : 227 xfer_align : 1 start_threshold : 1 stop_threshold : 15020 silence_threshold: 0 silence_size : 0 boundary : 2120220672
Takashi
participants (2)
-
Takashi Iwai -
William Juul