We want to stream audio from a Windows PC to a Raspberry PI over the network. First we tried to install ext13 with the External Manager on the PI but it has not worked, because there is no compiled version of ext13 for the ARM-Architecture the Raspberry PI uses. So we compiled ext13 on the PI accordingly to the README in the ext13 folder. Now we can create the streamin13~ object but nothing happens. Neither sound nor a error message. Have we forget something during compiling? In the README you can read that you have to put the compiled files to the pd directory. But where to put them exactly?
I hope someone can help us.

I don't know about [streamin13~] object, but there is a new solution recently which is "Audio over OSC" aka AoO (search for "aoo" in deken pakage).

https://git.iem.at/cm/aoo

Thank you! I will try this out.

@ecotec I use [udpreceive~] from the "net" library on my 64-bit rpi3. pretty sure compiled simply using "make", but it could have been added using help... Find externals.. (what is commonly called "Deken").
I had no luck with ext13.
David.

P.S....... set up like this......

We tested the aoo external. But now we have a lot of packet losses. Is this a general network problem or a Puredata specific one?

@ecotec Probably a general problem. The packets probably make it through the network, but can be delayed or arrive in the wrong order. UDP does not call back to check for data consistency so if a packet is dropped there will be a dropout.
Routers usually have a way to set QoS priority nowadays. If you set your router up for QoS for UDP on the ports you are using the data will be more reliable. QoS might be a sub-setting under VOIP.
The packet lengths need to be matched for the transmitter and receiver but hopefully AoO does that?
The packets are likely to be time stamped so the receiver should be able to re-order them if it has the time to do so...... setting a large enough buffer for the receiver would help with that....... if there is a mechanism for that.
TCP might be an option for AoO..... which would ensure packet arrival....... but a larger buffer would be required and so an increase in latency. That probably doesn't matter for streaming but it could be important for your actual requirements.
Windows processes audio at 48Khz. Does AoO do any resampling of audio if Pd is set to 44Khz? Look out for a de-tuned stream. I am unsure about whether that could be your next problem.
David.

Thank you very much for your detailed answer! I will check this options.