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

[Benno Senoner]

Steve Harris wrote:

>On Thu, Aug 12, 2004 at 07:32:02 -0400, Lee Revell wrote:
>  
>
>>On Thu, 2004-08-12 at 19:21, Jody McIntyre wrote:
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>>>On Thu, Aug 12, 2004 at 07:11:44PM -0400, Dave Robillard wrote:
>>>
>>>      
>>>
>>>>Well, once Jack has MIDI, all we need is a jack-over-ethernet driver,
>>>>and the world is ours! :)
>>>>
>>>>(Network transparent low latency sample-accurate cross-platform MIDI and
>>>>audio... mmmm...)
>>>>        
>>>>
>>>I assume you're joking, but I have to ask anyway: how are we to achieve
>>>low latencies over a medium with no bounds on transmit time? :)
>>>
>>>      
>>>
>>UDP.  If you don't get the packet in time, drop it and move on.
>>    
>>
>
>UDP also has unbounded transit time. In practice its OK if you dont want
>low latencies (just use RTP), but for low latency you really need one of
>the non-IP ethernet protocols that can be relaibly used for audio.
>  
>

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

>- Steve
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>