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Remove stdweb in favor of js-sys, web-sys, and wasm-bindgen (RustAudi…
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…o#713)
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@brightly-salty
brightly-salty authored Oct 25, 2022
1 parent 0965ead commit 29999e6
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Showing 4 changed files with 149 additions and 99 deletions.
5 changes: 4 additions & 1 deletion Cargo.toml
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Original file line number Diff line number Diff line change
Expand Up @@ -50,7 +50,10 @@ coreaudio-rs = { version = "0.10", default-features = false, features = ["audio_
coreaudio-rs = { version = "0.10", default-features = false, features = ["audio_unit", "core_audio", "audio_toolbox"] }

[target.'cfg(target_os = "emscripten")'.dependencies]
stdweb = { version = "0.4.20", default-features = false }
wasm-bindgen = { version = "0.2.58" }
wasm-bindgen-futures = "0.4.33"
js-sys = { version = "0.3.35" }
web-sys = { version = "0.3.35", features = [ "AudioContext", "AudioContextOptions", "AudioBuffer", "AudioBufferSourceNode", "AudioNode", "AudioDestinationNode", "Window", "AudioContextState"] }

[target.'cfg(all(target_arch = "wasm32", target_os = "unknown"))'.dependencies]
wasm-bindgen = { version = "0.2.58", optional = true }
Expand Down
209 changes: 113 additions & 96 deletions src/host/emscripten/mod.rs
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Original file line number Diff line number Diff line change
@@ -1,12 +1,9 @@
use std::mem;
use std::os::raw::c_void;
use std::slice::from_raw_parts;
use js_sys::Float32Array;
use std::time::Duration;
use stdweb;
use stdweb::unstable::TryInto;
use stdweb::web::set_timeout;
use stdweb::web::TypedArray;
use stdweb::Reference;
use wasm_bindgen::prelude::*;
use wasm_bindgen::JsCast;
use wasm_bindgen_futures::{spawn_local, JsFuture};
use web_sys::AudioContext;

use crate::traits::{DeviceTrait, HostTrait, StreamTrait};
use crate::{
Expand All @@ -30,9 +27,11 @@ pub struct Devices(bool);
#[derive(Clone, Debug, PartialEq, Eq)]
pub struct Device;

#[wasm_bindgen]
#[derive(Clone)]
pub struct Stream {
// A reference to an `AudioContext` object.
audio_ctxt_ref: Reference,
audio_ctxt: AudioContext,
}

// Index within the `streams` array of the events loop.
Expand All @@ -54,7 +53,6 @@ const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;

impl Host {
pub fn new() -> Result<Self, crate::HostUnavailable> {
stdweb::initialize();
Ok(Host)
}
}
Expand Down Expand Up @@ -186,7 +184,7 @@ impl DeviceTrait for Device {
config: &StreamConfig,
sample_format: SampleFormat,
data_callback: D,
error_callback: E,
_error_callback: E,
_timeout: Option<Duration>,
) -> Result<Self::Stream, BuildStreamError>
where
Expand All @@ -209,12 +207,8 @@ impl DeviceTrait for Device {
};

// Create the stream.
let audio_ctxt_ref = js!(return new AudioContext()).into_reference().unwrap();
let stream = Stream { audio_ctxt_ref };

// Specify the callback.
let mut user_data = (self, data_callback, error_callback);
let user_data_ptr = &mut user_data as *mut (_, _, _);
let audio_ctxt = AudioContext::new().expect("webaudio is not present on this system");
let stream = Stream { audio_ctxt };

// Use `set_timeout` to invoke a Rust callback repeatedly.
//
Expand All @@ -223,15 +217,12 @@ impl DeviceTrait for Device {
// See also: The call to `set_timeout` at the end of the `audio_callback_fn` which creates
// the loop.
set_timeout(
|| {
audio_callback_fn::<D, E>(
user_data_ptr as *mut c_void,
config,
sample_format,
buffer_size_frames,
)
},
10,
stream.clone(),
data_callback,
config,
sample_format,
buffer_size_frames as u32,
);

Ok(stream)
Expand All @@ -240,108 +231,141 @@ impl DeviceTrait for Device {

impl StreamTrait for Stream {
fn play(&self) -> Result<(), PlayStreamError> {
let audio_ctxt = &self.audio_ctxt_ref;
js!(@{audio_ctxt}.resume());
let future = JsFuture::from(
self.audio_ctxt
.resume()
.expect("Could not resume the stream"),
);
spawn_local(async {
match future.await {
Ok(value) => assert!(value.is_undefined()),
Err(value) => panic!("AudioContext.resume() promise was rejected: {:?}", value),
}
});
Ok(())
}

fn pause(&self) -> Result<(), PauseStreamError> {
let audio_ctxt = &self.audio_ctxt_ref;
js!(@{audio_ctxt}.suspend());
let future = JsFuture::from(
self.audio_ctxt
.suspend()
.expect("Could not suspend the stream"),
);
spawn_local(async {
match future.await {
Ok(value) => assert!(value.is_undefined()),
Err(value) => panic!("AudioContext.suspend() promise was rejected: {:?}", value),
}
});
Ok(())
}
}

// The first argument of the callback function (a `void*`) is a cast pointer to `self`
// and to the `callback` parameter that was passed to `run`.
fn audio_callback_fn<D, E>(
user_data_ptr: *mut c_void,
config: &StreamConfig,
sample_format: SampleFormat,
buffer_size_frames: usize,
) where
fn audio_callback_fn<D>(
mut data_callback: D,
) -> impl FnOnce(Stream, StreamConfig, SampleFormat, u32)
where
D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
E: FnMut(StreamError) + Send + 'static,
{
let num_channels = config.channels as usize;
let sample_rate = config.sample_rate.0;
let buffer_size_samples = buffer_size_frames * num_channels;

unsafe {
let user_data_ptr2 = user_data_ptr as *mut (&Stream, D, E);
let user_data = &mut *user_data_ptr2;
let (ref stream, ref mut data_cb, ref mut _err_cb) = user_data;
let audio_ctxt = &stream.audio_ctxt_ref;
|stream, config, sample_format, buffer_size_frames| {
let sample_rate = config.sample_rate.0;
let buffer_size_samples = buffer_size_frames * config.channels as u32;
let audio_ctxt = &stream.audio_ctxt;

// TODO: We should be re-using a buffer.
let mut temporary_buffer = vec![0f32; buffer_size_samples];
let mut temporary_buffer = vec![0f32; buffer_size_samples as usize];

{
let len = temporary_buffer.len();
let data = temporary_buffer.as_mut_ptr() as *mut ();
let mut data = Data::from_parts(data, len, sample_format);

let now_secs: f64 = js!(@{audio_ctxt}.getOutputTimestamp().currentTime)
.try_into()
.expect("failed to retrieve Value as f64");
let mut data = unsafe { Data::from_parts(data, len, sample_format) };
let now_secs: f64 = audio_ctxt.current_time();
let callback = crate::StreamInstant::from_secs_f64(now_secs);
// TODO: Use proper latency instead. Currently, unsupported on most browsers though, so
// we estimate based on buffer size instead. Probably should use this, but it's only
// supported by firefox (2020-04-28).
// let latency_secs: f64 = js!(@{audio_ctxt}.outputLatency).try_into().unwrap();
// let latency_secs: f64 = audio_ctxt.outputLatency.try_into().unwrap();
let buffer_duration = frames_to_duration(len, sample_rate as usize);
let playback = callback
.add(buffer_duration)
.expect("`playback` occurs beyond representation supported by `StreamInstant`");
let timestamp = crate::OutputStreamTimestamp { callback, playback };
let info = OutputCallbackInfo { timestamp };
data_cb(&mut data, &info);
data_callback(&mut data, &info);
}

// TODO: directly use a TypedArray<f32> once this is supported by stdweb
let typed_array = {
let f32_slice = temporary_buffer.as_slice();
let u8_slice: &[u8] = from_raw_parts(
f32_slice.as_ptr() as *const _,
f32_slice.len() * mem::size_of::<f32>(),
);
let typed_array: TypedArray<u8> = u8_slice.into();
typed_array
};

debug_assert_eq!(temporary_buffer.len() % num_channels as usize, 0);

js!(
var src_buffer = new Float32Array(@{typed_array}.buffer);
var context = @{audio_ctxt};
var buffer_size_frames = @{buffer_size_frames as u32};
var num_channels = @{num_channels as u32};
var sample_rate = sample_rate;

var buffer = context.createBuffer(num_channels, buffer_size_frames, sample_rate);
for (var channel = 0; channel < num_channels; ++channel) {
var buffer_content = buffer.getChannelData(channel);
for (var i = 0; i < buffer_size_frames; ++i) {
buffer_content[i] = src_buffer[i * num_channels + channel];
}
let typed_array: Float32Array = temporary_buffer.as_slice().into();

debug_assert_eq!(temporary_buffer.len() % config.channels as usize, 0);

let src_buffer = Float32Array::new(typed_array.buffer().as_ref());
let context = audio_ctxt;
let buffer = context
.create_buffer(
config.channels as u32,
buffer_size_frames as u32,
sample_rate as f32,
)
.expect("Buffer could not be created");
for channel in 0..config.channels {
let mut buffer_content = buffer
.get_channel_data(channel as u32)
.expect("Should be impossible");
for (i, buffer_content_item) in buffer_content.iter_mut().enumerate() {
*buffer_content_item =
src_buffer.get_index(i as u32 * config.channels as u32 + channel as u32);
}
}

var node = context.createBufferSource();
node.buffer = buffer;
node.connect(context.destination);
node.start();
);
let node = context
.create_buffer_source()
.expect("The buffer source node could not be created");
node.set_buffer(Some(&buffer));
context
.destination()
.connect_with_audio_node(&node)
.expect("Could not connect the audio node to the destination");
node.start().expect("Could not start the audio node");

// TODO: handle latency better ; right now we just use setInterval with the amount of sound
// data that is in each buffer ; this is obviously bad, and also the schedule is too tight
// and there may be underflows
set_timeout(
|| audio_callback_fn::<D, E>(user_data_ptr, config, sample_format, buffer_size_frames),
buffer_size_frames as u32 * 1000 / sample_rate,
1000 * buffer_size_frames as i32 / sample_rate as i32,
stream.clone().clone(),
data_callback,
&config,
sample_format,
buffer_size_frames as u32,
);
}
}

fn set_timeout<D>(
time: i32,
stream: Stream,
data_callback: D,
config: &StreamConfig,
sample_format: SampleFormat,
buffer_size_frames: u32,
) where
D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
{
let window = web_sys::window().expect("Not in a window somehow?");
window
.set_timeout_with_callback_and_timeout_and_arguments_4(
&Closure::once_into_js(audio_callback_fn(data_callback))
.dyn_ref::<js_sys::Function>()
.expect("The function was somehow not a function"),
time,
&stream.into(),
&((*config).clone()).into(),
&Closure::once_into_js(move || sample_format),
&buffer_size_frames.into(),
)
.expect("The timeout could not be set");
}

impl Default for Devices {
fn default() -> Devices {
// We produce an empty iterator if the WebAudio API isn't available.
Expand Down Expand Up @@ -377,14 +401,7 @@ fn default_output_device() -> Option<Device> {

// Detects whether the `AudioContext` global variable is available.
fn is_webaudio_available() -> bool {
stdweb::initialize();
js!(if (!AudioContext) {
return false;
} else {
return true;
})
.try_into()
.unwrap()
AudioContext::new().is_ok()
}

// Whether or not the given stream configuration is valid for building a stream.
Expand Down
31 changes: 29 additions & 2 deletions src/lib.rs
View file
Original file line number Diff line number Diff line change
Expand Up @@ -148,8 +148,12 @@
// Extern crate declarations with `#[macro_use]` must unfortunately be at crate root.
#[cfg(target_os = "emscripten")]
#[macro_use]
extern crate stdweb;
extern crate wasm_bindgen;
#[cfg(target_os = "emscripten")]
extern crate js_sys;
extern crate thiserror;
#[cfg(target_os = "emscripten")]
extern crate web_sys;

pub use error::*;
pub use platform::{
Expand All @@ -160,6 +164,8 @@ pub use samples_formats::{FromSample, Sample, SampleFormat, SizedSample, I24, I4
use std::convert::TryInto;
use std::ops::{Div, Mul};
use std::time::Duration;
#[cfg(target_os = "emscripten")]
use wasm_bindgen::prelude::*;

mod error;
mod host;
Expand All @@ -177,6 +183,7 @@ pub type OutputDevices<I> = std::iter::Filter<I, fn(&<I as Iterator>::Item) -> b
pub type ChannelCount = u16;

/// The number of samples processed per second for a single channel of audio.
#[cfg_attr(target_os = "emscripten", wasm_bindgen)]
#[derive(Debug, Clone, Copy, PartialEq, Eq, PartialOrd, Ord)]
pub struct SampleRate(pub u32);

Expand Down Expand Up @@ -210,15 +217,33 @@ pub type FrameCount = u32;
/// large, leading to latency issues. If low latency is desired, Fixed(BufferSize)
/// should be used in accordance with the SupportedBufferSize range produced by
/// the SupportedStreamConfig API.
#[derive(Clone, Debug, Eq, PartialEq)]
#[derive(Clone, Copy, Debug, Eq, PartialEq)]
pub enum BufferSize {
Default,
Fixed(FrameCount),
}

#[cfg(target_os = "emscripten")]
impl wasm_bindgen::describe::WasmDescribe for BufferSize {
fn describe() {}
}

#[cfg(target_os = "emscripten")]
impl wasm_bindgen::convert::IntoWasmAbi for BufferSize {
type Abi = wasm_bindgen::convert::WasmOption<u32>;
fn into_abi(self) -> Self::Abi {
match self {
Self::Default => None,
Self::Fixed(fc) => Some(fc),
}
.into_abi()
}
}

/// The set of parameters used to describe how to open a stream.
///
/// The sample format is omitted in favour of using a sample type.
#[cfg_attr(target_os = "emscripten", wasm_bindgen)]
#[derive(Clone, Debug, Eq, PartialEq)]
pub struct StreamConfig {
pub channels: ChannelCount,
Expand Down Expand Up @@ -267,6 +292,7 @@ pub struct SupportedStreamConfig {
///
/// Raw input stream callbacks receive `&Data`, while raw output stream callbacks expect `&mut
/// Data`.
#[cfg_attr(target_os = "emscripten", wasm_bindgen)]
#[derive(Debug)]
pub struct Data {
data: *mut (),
Expand Down Expand Up @@ -327,6 +353,7 @@ pub struct InputCallbackInfo {
}

/// Information relevant to a single call to the user's output stream data callback.
#[cfg_attr(target_os = "emscripten", wasm_bindgen)]
#[derive(Debug, Clone, PartialEq, Eq)]
pub struct OutputCallbackInfo {
timestamp: OutputStreamTimestamp,
Expand Down
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