This was added in the 1.13 update and I'm still not sure why. Other mods
seem to get away without it, so I think it's fine to remove.
Also remove the fake net manager, as that's part of Forge nowadays.
Fixes#1044.
- Fixes#1026
- The remaining bytes counter wasn't being decremented, so the code that
splits off smaller packets was unreachable. Thus all file slices were
being put into a single UploadFileMessage packet.
- Fix UpgradeSpeakerPeripheral not calling super.detach (so old
computers were never cleaned up)
- Correctly lock computer accesses inside SpeakerPeripheral
Fixes#1003.
Fingers crossed this is the last bug. Then I can bump the year and push
a new release tomorrow.
We're still a few days away from release, but don't think anything else
is going to change. And I /really/ don't want to have to write this
changelog (and then merge into later versions) on the 25th.
While Minecraft will automatically push a new buffer when one is
exhausted, this doesn't help if there's only a single buffer in the
queue, and you end up with stutter.
By enquing a buffer when receiving sound we ensure there's always
something queued. I'm not 100% happy with this solution, but it does
alleviate some of the concerns in #993.
Also reduce the size of the client buffer to 0.5s from 1.5s. This is
still enough to ensure seamless audio when the server is running slow (I
tested at 13 tps, but should be able to go much worse).
When the game is paused in SSP world, speakers are not ticked. However,
System.nanoTime() continues to increase, which means the next tick
speakers believe there has been a big jump and so schedule a bunch of
extra audio.
To avoid this, we keep track of how long the game has been paused offset
nanoTime by that amount.
Fixes#994
It's just more confusing having to keep track of where the ByteBuffer is
at. In this case, I think we were forgetting to rewind after computing
the digest.
Hopefully we'll be able to drop some of these in 1.17 as Java 16 has
a few more ByteBuffer methods
Fixes#992
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!
- Remove all the hungrarian notation in variables. Currently leaving
the format of rednet messages for now, while I work out whether this
counts as part of the public API or not.
- Fix the "repeat" program failing with broadcast packets. This was
introduced in #900, but I don't think anybody noticed. Will be more
relevant when #955 is implemented though.
