Closed oh3gai closed 5 years ago
FreeDV is a grate thing. If i have some time, I will test if sound piping from an SSB demod via UDP is a first step solution to get it running. A "bad" but working solution could be virtual audio cable. Native suport still would be best but on can only work at some thing at the same time.
It would be great if Codec2 or Opus could be used instead of AMBE for DMR or C4FM instead of mbelib.
Opus is now in the dependencies but used to send the audio stream through the network at 64 kbits/s. Opus has a minimal sample rate of 6 kbits/s. For best compatibility one should aim at 72 bits per 20 ms frame which sets the limit to 3.6 kbit/s. This is slightly above the upper limit of 3.2 kbit/s of Codec2 so one could imagine adding some FEC or slowing down the rate to improve robustness. Codec2 works still very well at 1.6 kbits/s. Maybe D-Star and GMSK would be the first thing to try. At half rate one could aim at a ~3kHz bandwidth. A quarter rate is even possible since it still works at 0.8 kbits/s. In order to conduct these experiments transmission support is necessary because of the lack of commercial equipment. A ticket has been open in the DSDcc project to this purpose.
The purpose of this one is FreeDV though. There's now a FreeDV API that can make implementation a lot easier.
Source code with API usage example: https://sourceforge.net/p/freetel/code/HEAD/tree/codec2-dev/
Thanks for the addition will test it in the coming days
Implemented in v4.5.0. A few things still need to be ironed out but will come later.
Hi!
First of all thanks for the awesome application. I was easily able to install it for both ubuntu and win7. Is it possible to integrate FreeDV/Codec2 to demodulators and sinks? I think this would have huge impact for popularity of amateur digital voice modes.
https://freedv.org/
Great work!
Br, Jarno