I need a system where the starting of the playback/recording has systematic (and minimal) delays.
Now I have read some faqs and, if I have understood right, it is not possible to have "systematic" delays and this all is the question of latencies. The system has 2.4.20-8 kernel without any low-latency-patches. BUT, there are so many latency patches out there, that I need some help to choose one - and it would be great if there was some howto-document too somewhere.
Another question is that does these patches really have effect on my system, where synchronisation between to computers is done using GPS (located too far from each other)?
When I do a test with two computers (with accurate GPS clocks) running two shell script: Script in computer 1: --------------------- STARTTIME1=`date +%y%m%d%H%M%S%N` echo 'Recording started at' $STARTTIME1 nice -n -20 arecord -D plughw:0,1 -t wav -f cd -d 60 $FILENAME STOPTIME1=`date +%y%m%d%H%M%S%N` echo 'Recording stopped at' $STOPTIME1
Script in computer 2: --------------------- STARTTIME2=`date +%y%m%d%H%M%S%N` echo 'Playback started at' $STARTTIME2 nice -n -20 aplay -D plughw:0,0,0 -f cd track_xx.wav STOPTIME2=`date +%y%m%d%H%M%S%N` echo 'Playback stopped at' $STOPTIME2
The scripts are launched from crontab and the output of the computer 2 is connected to input of computer 1. There are systematic delays in the variables STARTTIME2 and STOPTIME2 (within 1ms, good enough!) (the playback) and in the variable STARTTIME1. BUT when I look inside the captured file FILENAME, there is a random (!?!) delay within 500 ms in the audio and the variable STOPTIME2 has a random delay within 1000ms (doesn't correlate very well with the delay in the captured audio).
I wonder if there is something to do with the recording (ALSA 0.9.6., card RME Digi96/8 PST using ADAT optical input). Does the random delay have something to do with the initialization of the sound driver? Or is there a better way to test this delay?
Yours, panu
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