[PD] JACK affects UDP rate

Christof Ressi info at christofressi.com
Fri Jul 23 23:59:26 CEST 2021

On 23.07.2021 23:11, Roman Haefeli wrote:

> On Fri, 2021-07-23 at 21:52 +0200, Christof Ressi wrote:
>> When we overhauled the networking code, I noticed that the TCP and
>> UDP functions would both read up to N bytes (where N is currently
>> 4096) in a single recv() call. With TCP the buffer can contain
>> several FUDI (or other) messages, but with UDP it would only contain
>> a single packet. So UDP was effectively much more rate limited than
>> TCP.
> I think, it _does_ make sense to treat TCP and UDP differently. With
> TCP, you probably want to consume only as much as you can eat at the
> time while keeping the rest buffered for later. OTOH, it's questionable
> whether "surplus" UDP packets should be stored for later. Rather - I
> think - they should be simply discarded.

What do you mean with "surplus" UDP packets? It's totally natural that 
the UDP receive buffer contains more than one packet at a given time... 
Packets might arrive in bulks and/or several hosts can send to the same 
socket. Why should we intentionally discard them?

I think the fundamental problem is this:

We we want to

a) drain the UDP receive buffer as fast as possible to avoid packet loss

b) avoid blocking the audio thread by processing too many messages

If we only have a single thread (like in Pd), there's a conflict of 

A better design is to receive packets on a dedicated thread and put them 
on a (ideally unbounded) queue. The audio thread can then take packets 
and process them at its own pace. This is what the Supercollider Server 
does, for example. It is also how I personally write UDP server 
applications. https://github.com/pure-data/pure-data/pull/1261 also goes 
into this direction.


Back to TCP vs UDP: let's say you are sending 32-byte FUDI messages at a 
high rate. With the old behavior, TCP allowed to receive and dispatch 
128 messages in a row (4096 bytes in total), but UDP only allowed a 
single message. For me, this didn't make any sense. With the new 
behavior you get the same number of messages.


> It would be nice, if more than
> one packet could be received per tick, of course, but then to buffer
> could simply be flushed, so that only "fresh" packets are considered in
> the next tick. I _think_ that's what network devices do as well: send
> it now or forget it.
Sorry, I don't understand this paragraph at all...
> With blocksize=64, ideally one packet per tick is received. On macOS,
> it seems each tick without a packet delays the processing of the
> subsequent packets by one DSP block. After a few seconds on a bad
> connection (Wifi, for instance), the delay settles at 200-500ms and
> there is a very clean signal afterwards (which is not surprising with
> such a large buffer). On Linux, the latency stays in the range set by
> the receive buffer and late packets are perceived as drop out. It looks
> like not processed packets are flushed on Linux, but are not macOS.

Hmmm... usually incoming UDP packets are discarded if the UDP receive 
buffer is full. Are you saying that this is not the case on macOS? That 
would be very interesting. Maybe the buffer can automatically grow in 
size? Unfortunately, I couldn't find any information on this.

> Assuming the relevant poll functions you mentioned are the same for the
> same backend (JACK) on both platforms (macOS, Linux), why is the
> behavior still different?
I guess because of the different behavior/size of the UDP receive buffer.
> Thanks a lot, Christof, for your time and effort in explaining the
> details. You already helped a lot. I'm somewhat relieved that the issue
> has an explanation.
You're welcome :-)
> Roman
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