This adds SPDX license headers to all source code files, following the
REUSE[1] specification. This does not include any asset files (such as
generated JSON files, or textures). While REUSE does support doing so
with ".license" files, for now we define these licences using the
.reuse/dep5 file.
[1]: https://reuse.software/
Just ran[^1] over the codebase. Turns out we'd duplicated one of the
changelog entries entirely - I suspect due to a version merge gone
wrong!
[^1]: https://github.com/crate-ci/typos/
- Add a TOC to the Local IPs page.
- Increase the echo delay in our speaker audio page to 1.5s. This
sounds much better and is less clashy than 1s. Also add a
sleep(0) (eww, I know) to fix timeouts on some browsers/computers.
- Move Lua feature compat to a new "reference" section. Still haven't
figured out how to structure these docs - open to any ideas really.
- Mention FFmpeg as an option for converting to DFPWM (closes#1075).
- Allow data-mount to override built-in files. See my comment in #1069.
- Start making the summary lines for modules a little better. Just say
what the module does, rather than "The X API does Y" or "Provides Y".
There's still a lot of work to be done here.
- Bundle prism.js on the page, so we can highlight non-Lua code.
- Copy our local_ips wiki page to the main docs.
Speakers can now play arbitrary PCM audio, sampled at 48kHz and with a
resolution of 8 bits. Programs can build up buffers of audio locally,
play it using `speaker.playAudio`, where it is encoded to DFPWM, sent
across the network, decoded, and played on the client.
`speaker.playAudio` may return false when a chunk of audio has been
submitted but not yet sent to the client. In this case, the program
should wait for a speaker_audio_empty event and try again, repeating
until it works.
While the API is a little odd, this gives us fantastic flexibility (we
can play arbitrary streams of audio) while still being resilient in the
presence of server lag (either TPS or on the computer thread).
Some other notes:
- There is a significant buffer on both the client and server, which
means that sound take several seconds to finish after playing has
started. One can force it to be stopped playing with the new
`speaker.stop` call.
- This also adds a `cc.audio.dfpwm` module, which allows encoding and
decoding DFPWM1a audio files.
- I spent so long writing the documentation for this. Who knows if it'll
be helpful!
