Update the `emscripten` backend for the removal of the `EventLoop`

For the most part, behaviour should be largely unchanged, however each
individual stream now has its own `set_timeout` callback loop, rather
than using one for processing all streams at once.

Many TODOs remain within the `emscripten` backend. These were left
untouched for the most part in favour of addressing this in a more
web-focused, future PR.
This commit is contained in:
mitchmindtree 2019-12-14 11:50:57 +01:00
parent 70dcf2390a
commit 37d80b98cf
2 changed files with 189 additions and 258 deletions

View File

@ -1,7 +1,6 @@
use std::mem;
use std::os::raw::c_void;
use std::slice::from_raw_parts;
use std::sync::Mutex;
use stdweb;
use stdweb::Reference;
use stdweb::unstable::TryInto;
@ -17,25 +16,102 @@ use PauseStreamError;
use PlayStreamError;
use SupportedFormatsError;
use StreamData;
use StreamDataResult;
use StreamError;
use SupportedFormat;
use UnknownTypeOutputBuffer;
use traits::{DeviceTrait, EventLoopTrait, HostTrait, StreamIdTrait};
use traits::{DeviceTrait, HostTrait, StreamTrait};
// The emscripten backend currently works by instantiating an `AudioContext` object per `Stream`.
// Creating a stream creates a new `AudioContext`. Destroying a stream destroys it. Creation of a
// `Host` instance initializes the `stdweb` context.
/// The default emscripten host type.
#[derive(Debug)]
pub struct Host;
/// Content is false if the iterator is empty.
pub struct Devices(bool);
#[derive(Clone, Debug, PartialEq, Eq)]
pub struct Device;
pub struct Stream {
// A reference to an `AudioContext` object.
audio_ctxt_ref: Reference,
}
// Index within the `streams` array of the events loop.
#[derive(Debug, Clone, PartialEq, Eq, Hash)]
pub struct StreamId(usize);
pub type SupportedInputFormats = ::std::vec::IntoIter<SupportedFormat>;
pub type SupportedOutputFormats = ::std::vec::IntoIter<SupportedFormat>;
impl Host {
pub fn new() -> Result<Self, crate::HostUnavailable> {
stdweb::initialize();
Ok(Host)
}
}
impl Devices {
fn new() -> Result<Self, DevicesError> {
Ok(Self::default())
}
}
impl Device {
#[inline]
fn name(&self) -> Result<String, DeviceNameError> {
Ok("Default Device".to_owned())
}
#[inline]
fn supported_input_formats(&self) -> Result<SupportedInputFormats, SupportedFormatsError> {
unimplemented!();
}
#[inline]
fn supported_output_formats(&self) -> Result<SupportedOutputFormats, SupportedFormatsError> {
// TODO: right now cpal's API doesn't allow flexibility here
// "44100" and "2" (channels) have also been hard-coded in the rest of the code ; if
// this ever becomes more flexible, don't forget to change that
// According to https://developer.mozilla.org/en-US/docs/Web/API/BaseAudioContext/createBuffer
// browsers must support 1 to 32 channels at leats and 8,000 Hz to 96,000 Hz.
//
// UPDATE: We can do this now. Might be best to use `crate::COMMON_SAMPLE_RATES` and
// filter out those that lay outside the range specified above.
Ok(
vec![
SupportedFormat {
channels: 2,
min_sample_rate: ::SampleRate(44100),
max_sample_rate: ::SampleRate(44100),
data_type: ::SampleFormat::F32,
},
].into_iter(),
)
}
fn default_input_format(&self) -> Result<Format, DefaultFormatError> {
unimplemented!();
}
fn default_output_format(&self) -> Result<Format, DefaultFormatError> {
// TODO: because it is hard coded, see supported_output_formats.
Ok(
Format {
channels: 2,
sample_rate: ::SampleRate(44100),
data_type: ::SampleFormat::F32,
},
)
}
}
impl HostTrait for Host {
type Devices = Devices;
type Device = Device;
type EventLoop = EventLoop;
fn is_available() -> bool {
// Assume this host is always available on emscripten.
@ -53,15 +129,12 @@ impl HostTrait for Host {
fn default_output_device(&self) -> Option<Self::Device> {
default_output_device()
}
fn event_loop(&self) -> Self::EventLoop {
EventLoop::new()
}
}
impl DeviceTrait for Device {
type SupportedInputFormats = SupportedInputFormats;
type SupportedOutputFormats = SupportedOutputFormats;
type Stream = Stream;
fn name(&self) -> Result<String, DeviceNameError> {
Device::name(self)
@ -82,106 +155,87 @@ impl DeviceTrait for Device {
fn default_output_format(&self) -> Result<Format, DefaultFormatError> {
Device::default_output_format(self)
}
}
impl EventLoopTrait for EventLoop {
type Device = Device;
type StreamId = StreamId;
fn build_input_stream(
fn build_input_stream<D, E>(
&self,
device: &Self::Device,
format: &Format,
) -> Result<Self::StreamId, BuildStreamError> {
EventLoop::build_input_stream(self, device, format)
}
fn build_output_stream(
&self,
device: &Self::Device,
format: &Format,
) -> Result<Self::StreamId, BuildStreamError> {
EventLoop::build_output_stream(self, device, format)
}
fn play_stream(&self, stream: Self::StreamId) -> Result<(), PlayStreamError> {
EventLoop::play_stream(self, stream)
}
fn pause_stream(&self, stream: Self::StreamId) -> Result<(), PauseStreamError> {
EventLoop::pause_stream(self, stream)
}
fn destroy_stream(&self, stream: Self::StreamId) {
EventLoop::destroy_stream(self, stream)
}
fn run<F>(&self, callback: F) -> !
_format: &Format,
_data_callback: D,
_error_callback: E,
) -> Result<Self::Stream, BuildStreamError>
where
F: FnMut(Self::StreamId, StreamDataResult) + Send,
D: FnMut(StreamData) + Send + 'static,
E: FnMut(StreamError) + Send + 'static,
{
EventLoop::run(self, callback)
unimplemented!()
}
fn build_output_stream<D, E>(
&self,
_format: &Format,
data_callback: D,
error_callback: E,
) -> Result<Self::Stream, BuildStreamError>
where
D: FnMut(StreamData) + Send + 'static,
E: FnMut(StreamError) + Send + 'static,
{
// 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 (_, _, _);
// Use `set_timeout` to invoke a Rust callback repeatedly.
//
// The job of this callback is to fill the content of the audio buffers.
//
// 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), 10);
Ok(stream)
}
}
impl StreamIdTrait for StreamId {}
impl StreamTrait for Stream {
fn play(&self) -> Result<(), PlayStreamError> {
let audio_ctxt = &self.audio_ctxt_ref;
js!(@{audio_ctxt}.resume());
Ok(())
}
// The emscripten backend works by having a global variable named `_cpal_audio_contexts`, which
// is an array of `AudioContext` objects. A stream ID corresponds to an entry in this array.
//
// Creating a stream creates a new `AudioContext`. Destroying a stream destroys it.
// 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
pub struct EventLoop {
streams: Mutex<Vec<Option<Reference>>>,
fn pause(&self) -> Result<(), PauseStreamError> {
let audio_ctxt = &self.audio_ctxt_ref;
js!(@{audio_ctxt}.suspend());
Ok(())
}
}
impl EventLoop {
#[inline]
pub fn new() -> EventLoop {
stdweb::initialize();
EventLoop {
streams: Mutex::new(Vec::new()),
}
}
#[inline]
fn run<F>(&self, callback: F) -> !
where F: FnMut(StreamId, StreamDataResult),
{
// The `run` function uses `set_timeout` to invoke a Rust callback repeatidely. The job
// of this callback is to fill the content of the audio buffers.
// The first argument of the callback function (a `void*`) is a casted pointer to `self`
// and to the `callback` parameter that was passed to `run`.
fn callback_fn<F>(user_data_ptr: *mut c_void)
where F: FnMut(StreamId, StreamDataResult)
{
// The first argument of the callback function (a `void*`) is a casted 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)
where
D: FnMut(StreamData) + Send + 'static,
E: FnMut(StreamError) + Send + 'static,
{
unsafe {
let user_data_ptr2 = user_data_ptr as *mut (&EventLoop, F);
let user_data_ptr2 = user_data_ptr as *mut (&Stream, D, E);
let user_data = &mut *user_data_ptr2;
let user_cb = &mut user_data.1;
let streams = user_data.0.streams.lock().unwrap().clone();
for (stream_id, stream) in streams.iter().enumerate() {
let stream = match stream.as_ref() {
Some(v) => v,
None => continue,
};
let (ref stream, ref mut data_cb, ref mut _err_cb) = user_data;
let audio_ctxt = &stream.audio_ctxt_ref;
// TODO: We should be re-using a buffer.
let mut temporary_buffer = vec![0.0; 44100 * 2 / 3];
{
let buffer = UnknownTypeOutputBuffer::F32(::OutputBuffer { buffer: &mut temporary_buffer });
let data = StreamData::Output { buffer: buffer };
user_cb(StreamId(stream_id), Ok(data));
// TODO: directly use a TypedArray<f32> once this is supported by stdweb
data_cb(data);
}
// 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(
@ -197,7 +251,7 @@ impl EventLoop {
js!(
var src_buffer = new Float32Array(@{typed_array}.buffer);
var context = @{stream};
var context = @{audio_ctxt};
var buf_len = @{temporary_buffer.len() as u32};
var num_channels = @{num_channels};
@ -214,92 +268,11 @@ impl EventLoop {
node.connect(context.destination);
node.start();
);
}
set_timeout(|| callback_fn::<F>(user_data_ptr), 330);
}
}
let mut user_data = (self, callback);
let user_data_ptr = &mut user_data as *mut (_, _);
set_timeout(|| callback_fn::<F>(user_data_ptr as *mut _), 10);
stdweb::event_loop();
}
#[inline]
fn build_input_stream(&self, _: &Device, _format: &Format) -> Result<StreamId, BuildStreamError> {
unimplemented!();
}
#[inline]
fn build_output_stream(&self, _: &Device, _format: &Format) -> Result<StreamId, BuildStreamError> {
let stream = js!(return new AudioContext()).into_reference().unwrap();
let mut streams = self.streams.lock().unwrap();
let stream_id = if let Some(pos) = streams.iter().position(|v| v.is_none()) {
streams[pos] = Some(stream);
pos
} else {
let l = streams.len();
streams.push(Some(stream));
l
};
Ok(StreamId(stream_id))
}
#[inline]
fn destroy_stream(&self, stream_id: StreamId) {
self.streams.lock().unwrap()[stream_id.0] = None;
}
#[inline]
fn play_stream(&self, stream_id: StreamId) -> Result<(), PlayStreamError> {
let streams = self.streams.lock().unwrap();
let stream = streams
.get(stream_id.0)
.and_then(|v| v.as_ref())
.expect("invalid stream ID");
js!(@{stream}.resume());
Ok(())
}
#[inline]
fn pause_stream(&self, stream_id: StreamId) -> Result<(), PauseStreamError> {
let streams = self.streams.lock().unwrap();
let stream = streams
.get(stream_id.0)
.and_then(|v| v.as_ref())
.expect("invalid stream ID");
js!(@{stream}.suspend());
Ok(())
}
}
// Index within the `streams` array of the events loop.
#[derive(Debug, Clone, PartialEq, Eq, Hash)]
pub struct StreamId(usize);
// 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()
}
// Content is false if the iterator is empty.
pub struct Devices(bool);
impl Devices {
fn new() -> Result<Self, DevicesError> {
Ok(Self::default())
// 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), 330);
}
}
@ -336,54 +309,13 @@ fn default_output_device() -> Option<Device> {
}
}
#[derive(Clone, Debug, PartialEq, Eq)]
pub struct Device;
impl Device {
#[inline]
fn name(&self) -> Result<String, DeviceNameError> {
Ok("Default Device".to_owned())
}
#[inline]
fn supported_input_formats(&self) -> Result<SupportedInputFormats, SupportedFormatsError> {
unimplemented!();
}
#[inline]
fn supported_output_formats(&self) -> Result<SupportedOutputFormats, SupportedFormatsError> {
// TODO: right now cpal's API doesn't allow flexibility here
// "44100" and "2" (channels) have also been hard-coded in the rest of the code ; if
// this ever becomes more flexible, don't forget to change that
// According to https://developer.mozilla.org/en-US/docs/Web/API/BaseAudioContext/createBuffer
// browsers must support 1 to 32 channels at leats and 8,000 Hz to 96,000 Hz.
Ok(
vec![
SupportedFormat {
channels: 2,
min_sample_rate: ::SampleRate(44100),
max_sample_rate: ::SampleRate(44100),
data_type: ::SampleFormat::F32,
},
].into_iter(),
)
}
fn default_input_format(&self) -> Result<Format, DefaultFormatError> {
unimplemented!();
}
fn default_output_format(&self) -> Result<Format, DefaultFormatError> {
// TODO: because it is hard coded, see supported_output_formats.
Ok(
Format {
channels: 2,
sample_rate: ::SampleRate(44100),
data_type: ::SampleFormat::F32,
},
)
}
// 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()
}
pub type SupportedInputFormats = ::std::vec::IntoIter<SupportedFormat>;
pub type SupportedOutputFormats = ::std::vec::IntoIter<SupportedFormat>;

View File

@ -435,9 +435,8 @@ mod platform_impl {
pub use crate::host::emscripten::{
Device as EmscriptenDevice,
Devices as EmscriptenDevices,
EventLoop as EmscriptenEventLoop,
Host as EmscriptenHost,
StreamId as EmscriptenStreamId,
Stream as EmscriptenStream,
SupportedInputFormats as EmscriptenSupportedInputFormats,
SupportedOutputFormats as EmscriptenSupportedOutputFormats,
};