[linux-audio-dev] Audio synchronization, MIDI API

Steve Harris wrote: I don't think raw ethernet will buy us anything over using UDP. These few usecs less simply won't matter. (but with ethernet you would have the disadvantage that you loose routability) On a 100Mbit network the round trip latency between hosts is about 100usecs so the one way latency of MIDI would be about half of that. and that's form a MIDI point of view instantaneous because over serial MIDI cable transmitting a NOTE ON event (3 bytes) takes about 1.1msec which is 20 times slower than transmitting it over an ethernet cable. About the packet loss ratio: besides from broken equipement (avoid hubs completely, use only switched networks), the packet loss ration in a LAN is pratically zero. What you must avoid is to saturate the network (eg having 2 hosts that flood the 100Mbit network with over 100Mbit) because at that point packets might get lost. But only a fool would try to use a saturated network to do professional audio/midi work and hoping to achieve low latencies and no packet loss. I tried a cheap $50 8port 100Mbit D-link switch and sent udp packets as fast as possible from one linux box to the other and measured the packet loss ratio (or on host1 you can use ping -f host2). It ran several hours and the packet loss ratio was zero. This means you can use UDP for audio/midi on a LAN without fearing compromising your audio work. Of course all elements must play nice (all machines must run on low latency kernels and low latency friendly audio cards using reasonable buffer sizes). If you follow a few simple rules audio/midi over UDP is just as reliable as a dedicated analogue cable. RTP is simply UDP with statistic gathering which could be useful in some cases but not necessarily needed. The onlyproblem I envision if you want to synchronize multiple audio cards over the network. In that case I guess sample skipping/duplication (from time to time) is the way to go. It's not hard to do but the hard thing is to keep buffers small and have the algorithm react fast to eventual drifts in sample rate. I measured the sample rate of some consumer (44100Hz) audio cards and usually it is quite stable but it varies 3-4 samples from card to card. eg card1 had 44096Hz , card2 44098Hz so assuming we use 128 frames buffers and the samplerate delta between cards is 2 samples we must insert/skip 2 samples per second in one of the cards. which means insert/skip a sample in one of each 172 buffers (of 128frames each) processed. Any good suggestion how to best implement it ? (if the audio slaves are without audio card then they will synchronize to the main node having the audio card which is easy to do) cheers, Benno http://www.linuxsampler.org