28 May
2017
28 May
'17
8:49 p.m.
Hi,
I think audio over wi-fi is a non-starter. it will never be glitch free.
I’m rewiring my own studio at the moment with fast router and cat 7 ethernet.
I use Artsmesh (MacOS) to route all the audio/video in the studio and over the internet
(uses jack/jacktrip, ffmpeg/syphon) and has a better interface than qjackctl.
Ken
On May 29, 2017, at 7:47 AM, Chris Caudle
<chris(a)chriscaudle.org> wrote:
On Tue, May 23, 2017 12:47 pm, Malik Costet wrote:
On 2017-05-23 18:22, Hanspeter Portner wrote:
>> ...But it only works great over Ethernet. And I'd very much like it to
>> work over WLAN, which it doesn't. Over wireless, the sound is
>> completely garbled. AFAICT, it's a network issue.
>
> NetJACK sends its audio data via UDP [1], e.g. there's no resending of
> lost
> packets like in TCP. If packets are lost (which is *very* likely over
> wireless), the packets are really lost and you miss a chunk of audio
> frames, hence your garbled sound at the receiver.
So your view is that packet loss is the cause? I
had naively assumed it
might be due to latency or congestion, but packet loss would make a lot
of sense. Although -- I hadn't mentioned this explicitly, but my setup
is with both boxes connected on the same local network, in other words
both to the same WiFi router; can there really be such a dramatic amount
of packet loss in that context for the result to be completely garbled
sound? I'm not talking about clicks here and there, but really little
better than static.
If the sound is always garbled, it never even starts out with decent
quality, then you may be right, latency could be causing every cycle to
not have enough time. Do you get a lot of under-run notifications when
starting? If latency is the only problem you may be able to set a larger
number of network cycles and get around that.
You should be able to get some packet loss statistics from ifconfig, it
would be worth checking to see if there are lots of RX errors or RX
dropped notifications. I don't know how well that information is gathered
for UDP packets, so using iperf to generate traffic and then looking how
many dropped packets were detected would be good. You could probably get
an idea from iperf of the average and maximum latency as well.
In general real time audio transmission over Wi-Fi is going to be
difficult. I have no trouble streaming FLAC files to music players over
Wi-Fi, but those buffer over half a second of audio, and are using TCP so
the TCP layer will just send again any packets which get dropped. When
using jack the network connection has to be able to send packets reliably
with low latency, and no matter what you do Wi-Fi is still going to be a
shared medium, so every time another device anywhere in the local vicinity
transmits, your devices are going to have to listen to see if they need to
switch to a different channel, or pause to allow fair time for another
device. That other device might belong to your neighbor, so you can't
assume that just because you have only two devices connected to your
router that there will not be other interference.
--
Chris Caudle
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