JAVA RECORDING PROBLEMS with 8 16 24 and 48k

[makaf]

Dear Matt,

i am struggeling when recording with a JAVA-application at different rates.

first i checked with aplay and arecord and it worked fine.

when trying the same through a JAVA application only 24k works, all other rates do not start and show same error (pls see attachement #1).
if i use the anyrate-alsa-setup (described in other post) there is no error but sound is distorted.

i attached small java-routine where you can make out the audioinjector is recognized correctly by JAVA but if you start sending the errors come up.

it would be great if you could suggest a workaround for this situation

best regards
markus

p.s. alsa test-setup

#!/bin/bash
sudo arecord -v -d10 -Dplughw:0,0 -fS16_LE -r 8000 -c2 test8k.wav
sudo arecord -v -d10 -Dplughw:0,0 -fS16_LE -r 16000 -c2 test16k.wav
sudo arecord -v -d10 -Dplughw:0,0 -fS16_LE -r 24000 -c2 test24k.wav
sudo arecord -v -d10 -Dplughw:0,0 -fS16_LE -r 48000 -c2 test48k.wav
aplay -v -Dplughw:0,0 test8k.wav
aplay -v -Dplughw:0,0 test16k.wav
aplay -v -Dplughw:0,0 test32k.wav
aplay -v -Dplughw:0,0 test48k.wav

#/etc/asound.conf
#pcm.!default {
# type asym
# playback.pcm "dmixplug"
# capture.pcm "dsnoopplug"
# }
# pcm.dmixplug {
# type plug slave.pcm "dmix"
# }
# pcm.dsnoopplug {
# type plug slave.pcm "dsnoop"
# }

[flatmax]

ok - it seems that the plughw is working, which suggests it is a problem to do with the specific hw formats you are requesting.
Can you please try equivalent hw formats as wav files. You can use sox to generate them.

sox inputFile -r rate -c channels -b .... and so on ... outputFile

Then aplay to test.

Matt

[makaf]

hi Matt,

pls see results with sox where only the 24k is dropping to 22050.
i will chk with ffmpeg and different java-runtime versions next.

best regards
markus


SOX RECORD MP3
Sample Rate : 16000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
sox WARN formats: can't set sample rate 24000; using 22050
Sample Rate : 22050 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
Sample Rate : 32000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
SOX PLAY MP3
sox_16k.mp3: Samplerate: 16000Hz
sox_24k.mp3: Samplerate: 22050Hz
sox_32k.mp3: Samplerate: 32000Hz
sox_48k.mp3: Samplerate: 48000Hz
SOX RECORD FLAC
Channels : 2 Sample Rate : 16000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
sox WARN formats: can't set sample rate 24000; using 22050
Channels : 2 Sample Rate : 22050 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
Channels : 2 Sample Rate : 32000 Precision : 16-bit Signed Integer PCM
Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM
SOX PLAY FLAC
sox_16k.flac: Encoding: FLAC Info: Processed by SoX Channels: 2 @ 16-bit Samplerate: 16000Hz
sox_24k.flac: Encoding: FLAC Info: Processed by SoX Channels: 2 @ 16-bit Samplerate: 22050Hz
sox_32k.flac: Encoding: FLAC Info: Processed by SoX Channels: 2 @ 16-bit Samplerate: 32000Hz
sox_48k.flac: Encoding: FLAC Info: Processed by SoX Channels: 2 @ 16-bit Samplerate: 48000Hz

ok - it seems that the plughw is working, which suggests it is a problem to do with the specific hw formats you are requesting.
Can you please try equivalent hw formats as wav files. You can use sox to generate them.

sox inputFile -r rate -c channels -b .... and so on ... outputFile

Then aplay to test.

Matt

[makaf]

dear Matt,

pls see the results with ffmpeg are the same as with sox before.

do not get confused because i had to continue testing with your OCTO-soundcard, so the channels are different.

br
markus


# ffmpeg -t 15 -f alsa -ac 8 -ar 16000 -i hw:0,0 -f flac ffmpeg16k.flac
Stream #0:0: Audio: pcm_s16le, 16000 Hz, 7.1, s16, 2048 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))
Stream #0:0: Audio: pcm_s16le, 22050 Hz, 7.1, s16, 2822 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))
Stream #0:0: Audio: pcm_s16le, 32000 Hz, 7.1, s16, 4096 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))
Stream #0:0: Audio: pcm_s16le, 48000 Hz, 7.1, s16, 6144 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> flac (native))

play ffmpeg*.flac
ffmpeg16k.flac: Encoding: FLAC Channels: 8 @ 16-bit Samplerate: 16000Hz
ffmpeg24k.flac: Encoding: FLAC Channels: 8 @ 16-bit Samplerate: 22050Hz
ffmpeg32k.flac: Encoding: FLAC Channels: 8 @ 16-bit Samplerate: 32000Hz
ffmpeg48k.flac: Encoding: FLAC Channels: 8 @ 16-bit Samplerate: 48000Hz

[flatmax]

Regarding 24kHz on the stereo card, I think it only supports these rates :
8000, 16000, 32000, 44100, 48000, 96000, 88200,

due to recent changes with the i2s driver in the kernel, I need to update the machine driver to get 8kHz back.

Matt

[makaf]

dear Matt,

case closed

we found wrong settings in octave which caused the issues.

24 and 48k are tested and working fine

thank you

best regards
markus