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Cleanup

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This commit is contained in:
restitux
2024-05-25 18:01:01 -06:00
parent cf6458aae9
commit df6e0aa697
1 changed files with 20 additions and 101 deletions
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src/lib.rs
+20 -101
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@@ -1,39 +1,27 @@
pub mod app;
use std::cell::RefCell;
use std::pin::pin;
use std::rc::Rc;
use std::time::Duration;
use app::STATE;
use asynchronous_codec::Decoder;
use asynchronous_codec::Encoder;
use futures::AsyncRead;
use futures::AsyncWrite;
use futures::Sink;
use futures::SinkExt;
use futures::Stream;
use futures::StreamExt;
use mumble_protocol::control::ControlCodec;
use mumble_protocol::control::ControlPacket;
use mumble_protocol::control::{msgs, ClientControlCodec};
use mumble_protocol::Clientbound;
use mumble_protocol::Serverbound;
use wasm_bindgen::prelude::*;
use wasm_bindgen_futures::JsFuture;
use web_sys::console;
use web_sys::js_sys::Uint8Array;
use web_sys::AudioData;
use web_sys::AudioContext;
use web_sys::AudioContextOptions;
use web_sys::MediaStream;
use web_sys::AudioData;
use web_sys::AudioDecoder;
use web_sys::AudioDecoderConfig;
use web_sys::AudioDecoderInit;
use web_sys::CodecState;
use web_sys::EncodedAudioChunk;
use web_sys::EncodedAudioChunkInit;
use web_sys::EncodedAudioChunkType;
use web_sys::MediaStream;
use web_sys::MediaStreamTrackGenerator;
use web_sys::MediaStreamTrackGeneratorInit;
use web_sys::WebTransport;
@@ -41,13 +29,9 @@ use web_sys::WebTransportOptions;
use wasm_bindgen_futures::spawn_local as spawn;
//#[wasm_bindgen]
//extern "C" {
//}
fn configure_audio_context(
audio_stream_generator: &MediaStreamTrackGenerator,
) -> AudioContext {
// Borrowed from
// https://github.com/security-union/videocall-rs/blob/main/videocall-client/src/decode/config.rs#L6
fn configure_audio_context(audio_stream_generator: &MediaStreamTrackGenerator) -> AudioContext {
let js_tracks = web_sys::js_sys::Array::new();
js_tracks.push(audio_stream_generator);
let media_stream = MediaStream::new_with_tracks(&js_tracks).unwrap();
@@ -67,6 +51,9 @@ fn configure_audio_context(
}
pub async fn network_entrypoint() {
// This sleep is to allow the user to interact with the window so the MediaStream
// can be created. This works around Chrome's autoplay policy rules. This will
// eventually be unnecessary when we have a proper GUI.
async_std::task::sleep(Duration::from_millis(3000)).await;
console::log_1(&"Rust via WASM!".into());
@@ -177,6 +164,7 @@ pub async fn network_entrypoint() {
send_chan.send(msg.into()).await.unwrap();
console::log_1(&"Sent authenticate packet".into());
// Spawn worker to send pings
{
let send_chan = send_chan.clone();
spawn(async move {
@@ -192,10 +180,12 @@ pub async fn network_entrypoint() {
});
}
// Create callback functions for AudioDecoder
let error = Closure::wrap(Box::new(move |e: JsValue| {
console::log_1(&e);
}) as Box<dyn FnMut(JsValue)>);
// Create MediaStreams to playback decoded audio
let audio_stream_generator =
MediaStreamTrackGenerator::new(&MediaStreamTrackGeneratorInit::new("audio")).unwrap();
// The audio context is used to reproduce audio.
@@ -226,17 +216,13 @@ pub async fn network_entrypoint() {
let audio_decoder = AudioDecoder::new(&AudioDecoderInit::new(
error.as_ref().unchecked_ref(),
output.as_ref().unchecked_ref(),
)).unwrap();
))
.unwrap();
console::log_1(&"Created Audio Decoder".into());
console::log_1(&audio_decoder);
audio_decoder.configure(&AudioDecoderConfig::new(
"opus",
1,
48000,
));
audio_decoder.configure(&AudioDecoderConfig::new("opus", 1, 48000));
loop {
match reader.next().await {
@@ -259,28 +245,15 @@ pub async fn network_entrypoint() {
) = payload
{
let js_audio_payload = Uint8Array::from(audio_payload.as_ref());
audio_decoder.decode(&EncodedAudioChunk::new(
&EncodedAudioChunkInit::new(
audio_decoder.decode(
&EncodedAudioChunk::new(&EncodedAudioChunkInit::new(
&js_audio_payload.into(),
0.0,
EncodedAudioChunkType::Key,
),
).unwrap());
))
.unwrap(),
);
console::log_1(&"Oueued audio chunk for decoding".into());
//let mut encoded_audio_chunk_init =
// web_sys::EncodedAudioChunkInit::new(
// &js_audio_payload.into(),
// 0.0,
// web_sys::EncodedAudioChunkType::Delta,
// );
////encoded_audio_chunk_init.duration(1.0);
//let encoded_audio_chunk =
// web_sys::EncodedAudioChunk::new(&encoded_audio_chunk_init)
// .unwrap();
//console::log_1(&encoded_audio_chunk);
}
}
_ => {
@@ -305,58 +278,4 @@ pub async fn network_entrypoint() {
}
}
}
//async fn handle_send(
// mut writer: Rc<RefCell<asynchronous_codec::FramedWrite<impl AsyncRead + AsyncWrite + Unpin, ControlCodec<Serverbound, Clientbound>>>>,
// send_queue: Queue<mumble_protocol::Serverbound>,
//) {
// loop {
// let msg = send_queue.get();
// client.send(msg).await.unwrap();
// }
//}
//async fn handle_ping(
//) {
// //pin!(client);
// loop {
// let ping = msgs::Ping::new();
// client.borrow_mut().send(ping.into()).await.unwrap();
// console::log_1(&"Sent ping packet".into());
//
// async_std::task::sleep(Duration::from_millis(3000)).await;
// }
//}
//let queue_write, queue_read = Queue::new();
//spawn(handle_send(writer, queue_write));
//spawn(handle_ping(queue_read));
//loop {
// let msg = reader.next().await.unwrap().unwrap();
// console::log_1(&format!("{:#?}", msg).into());
//}
// let result: Option<(Version, u32)> = loop {
// match client.next().await {
// None => break None,
// Some(packet) => {
// let packet = packet.unwrap();
// }
// }
// };
//let client = ClientControlCodec::new().framed(stream);
//let client = ClientControlCodec::new().framed(stream);
//let reader = stream.readable();
//let writer = stream.writable();
//console::log_1(&stream.into());
//*STATE.status.write() = "Ready!".into();
//return stream.into();
}