Harold,

"Depending on the desired frequency, each successive sample would be some
number of steps through a sine table from the previous sample."

I think that would work as long as the sin wave table was large enough to avoid aliasing issues.
Also would need to be done on something like an Arduino NOT on a multi-tasking OS like RPi.
Interrupts would interfere with the timing.

By "compression" I think you are referring to data compression (bandwidth) not audio compression. (volume dynamics)
If I understand correctly formats like MP3 take less bandwidth by using some sort of compression algorithm and other formats like WAV are just strait up PCM resulting in larger files.
Same with phone codecs.
Some are better than others. Not all will pass FAX data.
I think the codec on my phone is set to pass every other word. lol

-Steve


On Thursday, September 23, 2021, 12:00:34 PM EDT, Harold Hallikainen <[email protected]> wrote:


Methods of doing AFSK Internet broadcasts like ITTY are interesting. I
thought the idea of splicing together mark and space tone files was very
clever.

I wonder about the need for audio compression. I did a microcontroller
project that streamed live audio using ADPCM and PCM. The PCM used 16 bit
samples at 44.41 kHz. But, I think for RTTY, 8 bit samples at 8 kHz should
be fine (since that's what circuit switched telephone uses). In my
application, on receiving the HTTP request, I sent the WAV header with a
very large file size, then just started sending the samples.

As mentioned, I do think the splicing of tone files is very clever!
Another method that would be interesting is direct digital synthesis.
Depending on the desired frequency, each successive sample would be some
number of steps through a sine table from the previous sample.

Anyway, interesting stuff!

Harold


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