Rate conversion on Linux

[w4mmp-ron]

Hello,
I am revising (updating) an SDR application that runs on the RPi 4/5 under Raspbian O.S. What if anything provides rate conversion as Windows does? Does PortAudio now support rate conversion. If not, what does? The SDR interfaces with a transceiver that only supports 2 channel, 16bit, 96K.

Regards,
Ron

[philburk]

PortAudio does not have its own rate conversion code.
But there is some handy C++ code for a resampler on GitHub that can be used under the AOSP license.
https://github.com/google/oboe/tree/main/src/flowgraph/resampler

[RossBencina]

I'm not Linux audio expert. PortAudio only implements sample rate conversion if the underlying host API implements it. On Linux this may well be available transparently with some host APIs and/or some devices. I believe that it is possible for the user to configure a kind of virtual ALSA device that wraps the hardware device and provides sample rate conversion. [EDIT: according to Claude, opening "plughw:0,0" instead of "hw:0,0" gets you sample rate conversion for free.] So I think the first thing for you to do is to research what the standard way to handle this on Linux is. Is it the user's responsibility or the client software's responsibility.

That said, for an SDR application I think you should take the data exactly as the transceiver provides it and do your own DSP as required.

[w4mmp-ron]

Hi,
Thanks to both of you for getting back to me. I want to thank everyone involved in producing the most recent version of portaudio. While rate conversion is still a bit of any issue for me, the new APIs are great! The support of Pipewire/PulseAudio did solve one of my long outstanding issues.
Thanks for the resampler link. I will have Grok/Build take a look and help determine if it can help solve the other issue of resampling.
Cheers