nandi/jolt-native · mod.rs

⑂ 2ab40bf193b6a39bc8d1397cc36d17572bbf59cd

mod.rs · 439 lines · 14.1 KBRust    Blame HistoryRaw
Lift freeq's AV media plane out of sleek   1//! Emscripten backend implementation.
3//! Default backend on Emscripten.
5use js_sys::Float32Array;
6use std::panic::AssertUnwindSafe;
7use std::time::Duration;
8use wasm_bindgen::prelude::*;
9use wasm_bindgen::JsCast;
10use wasm_bindgen_futures::{spawn_local, JsFuture};
11use web_sys::AudioContext;
13use crate::traits::{DeviceTrait, HostTrait, StreamTrait};
14use crate::{
  BufferSize, BuildStreamError, Data, DefaultStreamConfigError, DeviceDescription,
  DeviceDescriptionBuilder, DeviceId, DeviceIdError, DeviceNameError, DevicesError,
  InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleFormat,
  SampleRate, StreamConfig, StreamError, SupportedBufferSize, SupportedStreamConfig,
  SupportedStreamConfigRange, SupportedStreamConfigsError,
22// The emscripten backend currently works by instantiating an `AudioContext` object per `Stream`.
23// Creating a stream creates a new `AudioContext`. Destroying a stream destroys it. Creation of a
24// `Host` instance initializes the `stdweb` context.
26/// The default emscripten host type.
27#[derive(Debug)]
28pub struct Host;
30/// Content is false if the iterator is empty.
31pub struct Devices(bool);
33#[derive(Clone, Debug, PartialEq, Eq)]
34pub struct Device;
36#[wasm_bindgen]
37#[derive(Clone)]
38pub struct Stream {
  // A reference to an `AudioContext` object.
  audio_ctxt: AudioContext,
43// WASM runs in a single-threaded environment, so Send and Sync are safe by design.
44unsafe impl Send for Stream {}
45unsafe impl Sync for Stream {}
47// Compile-time assertion that Stream is Send and Sync
48crate::assert_stream_send!(Stream);
49crate::assert_stream_sync!(Stream);
51pub use crate::iter::{SupportedInputConfigs, SupportedOutputConfigs};
53const MIN_CHANNELS: u16 = 1;
54const MAX_CHANNELS: u16 = 32;
55const MIN_SAMPLE_RATE: SampleRate = 8_000;
56const MAX_SAMPLE_RATE: SampleRate = 96_000;
57const DEFAULT_SAMPLE_RATE: SampleRate = 44_100;
58const MIN_BUFFER_SIZE: u32 = 1;
59const MAX_BUFFER_SIZE: u32 = u32::MAX;
60const DEFAULT_BUFFER_SIZE: usize = 2048;
61const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;
63impl Host {
  pub fn new() -> Result<Self, crate::HostUnavailable> {
      Ok(Host)
69impl Devices {
  fn new() -> Result<Self, DevicesError> {
      Ok(Self::default())
75impl Device {
  fn description(&self) -> Result<DeviceDescription, DeviceNameError> {
      Ok(DeviceDescriptionBuilder::new("Default Device".to_string())
          .direction(crate::DeviceDirection::Output)
          .build())
  fn id(&self) -> Result<DeviceId, DeviceIdError> {
      Ok(DeviceId(
          crate::platform::HostId::Emscripten,
          "default".to_string(),
  fn supported_input_configs(
      &self,
  ) -> Result<SupportedInputConfigs, SupportedStreamConfigsError> {
      unimplemented!();
  fn supported_output_configs(
      &self,
  ) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
      let buffer_size = SupportedBufferSize::Range {
          min: MIN_BUFFER_SIZE,
          max: MAX_BUFFER_SIZE,
      let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
          .map(|channels| SupportedStreamConfigRange {
              channels,
              min_sample_rate: MIN_SAMPLE_RATE,
              max_sample_rate: MAX_SAMPLE_RATE,
              buffer_size,
              sample_format: SUPPORTED_SAMPLE_FORMAT,
          .collect();
      Ok(configs.into_iter())
  fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
      unimplemented!();
  fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
      const EXPECT: &str = "expected at least one valid webaudio stream config";
      let config = self
          .supported_output_configs()
          .expect(EXPECT)
          .max_by(|a, b| a.cmp_default_heuristics(b))
          .unwrap()
          .with_sample_rate(DEFAULT_SAMPLE_RATE);
      Ok(config)
131impl HostTrait for Host {
  type Devices = Devices;
  type Device = Device;
  fn is_available() -> bool {
      // Assume this host is always available on emscripten.
      true
  fn devices(&self) -> Result<Self::Devices, DevicesError> {
      Devices::new()
  fn default_input_device(&self) -> Option<Self::Device> {
      default_input_device()
  fn default_output_device(&self) -> Option<Self::Device> {
      default_output_device()
153impl DeviceTrait for Device {
  type SupportedInputConfigs = SupportedInputConfigs;
  type SupportedOutputConfigs = SupportedOutputConfigs;
  type Stream = Stream;
  fn description(&self) -> Result<DeviceDescription, DeviceNameError> {
      Device::description(self)
  fn id(&self) -> Result<DeviceId, DeviceIdError> {
      Device::id(self)
  fn supported_input_configs(
      &self,
  ) -> Result<Self::SupportedInputConfigs, SupportedStreamConfigsError> {
      Device::supported_input_configs(self)
  fn supported_output_configs(
      &self,
  ) -> Result<Self::SupportedOutputConfigs, SupportedStreamConfigsError> {
      Device::supported_output_configs(self)
  fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
      Device::default_input_config(self)
  fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
      Device::default_output_config(self)
  fn build_input_stream_raw<D, E>(
      &self,
      _config: StreamConfig,
      _sample_format: SampleFormat,
      _data_callback: D,
      _error_callback: E,
      _timeout: Option<Duration>,
  ) -> Result<Self::Stream, BuildStreamError>
194
      D: FnMut(&Data, &InputCallbackInfo) + Send + 'static,
      E: FnMut(StreamError) + Send + 'static,
      unimplemented!()
  fn build_output_stream_raw<D, E>(
      &self,
      config: StreamConfig,
      sample_format: SampleFormat,
      data_callback: D,
      _error_callback: E,
      _timeout: Option<Duration>,
  ) -> Result<Self::Stream, BuildStreamError>
209
      D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
      E: FnMut(StreamError) + Send + 'static,
      if !valid_config(config, sample_format) {
          return Err(BuildStreamError::StreamConfigNotSupported);
      let buffer_size_frames = match config.buffer_size {
          BufferSize::Fixed(v) => {
              if !(MIN_BUFFER_SIZE..=MAX_BUFFER_SIZE).contains(&v) {
                  return Err(BuildStreamError::StreamConfigNotSupported);
              v as usize
          BufferSize::Default => DEFAULT_BUFFER_SIZE,
      // Create the stream.
      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.
      // 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.
      let data_callback = AssertUnwindSafe(data_callback);
      set_timeout(
          stream.clone(),
          data_callback,
          config,
          sample_format,
          buffer_size_frames as u32,
      Ok(stream)
251impl StreamTrait for Stream {
  fn play(&self) -> Result<(), PlayStreamError> {
      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 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(())
283fn audio_callback_fn<D>(
  mut data_callback: AssertUnwindSafe<D>,
285) -> impl FnOnce(Stream, StreamConfig, SampleFormat, u32)
  D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
  |stream, config, sample_format, buffer_size_frames| {
      let sample_rate = config.sample_rate;
      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 as usize];
          let len = temporary_buffer.len();
          let data = temporary_buffer.as_mut_ptr() as *mut ();
          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 = 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_callback(&mut data, &info);
      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,
              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);
      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(
          1000 * buffer_size_frames as i32 / sample_rate as i32,
          stream.clone().clone(),
          data_callback,
          config,
          sample_format,
          buffer_size_frames,
363fn set_timeout<D>(
  time: i32,
  stream: Stream,
  data_callback: AssertUnwindSafe<D>,
  config: StreamConfig,
  sample_format: SampleFormat,
  buffer_size_frames: u32,
  D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
  let window = web_sys::window().expect("Not in a window somehow?");
374
      .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.into(),
          &Closure::once_into_js(move || sample_format),
          &buffer_size_frames.into(),
      .expect("The timeout could not be set");
388impl Default for Devices {
  fn default() -> Devices {
      // We produce an empty iterator if the WebAudio API isn't available.
      Devices(is_webaudio_available())
394impl Iterator for Devices {
  type Item = Device;
  fn next(&mut self) -> Option<Device> {
      if self.0 {
          self.0 = false;
          Some(Device)
      } else {
          None
407fn default_input_device() -> Option<Device> {
  unimplemented!();
411fn default_output_device() -> Option<Device> {
  if is_webaudio_available() {
      Some(Device)
  } else {
      None
419// Detects whether the `AudioContext` global variable is available.
420fn is_webaudio_available() -> bool {
  AudioContext::new().is_ok()
424// Whether or not the given stream configuration is valid for building a stream.
425fn valid_config(conf: StreamConfig, sample_format: SampleFormat) -> bool {
  conf.channels <= MAX_CHANNELS
      && conf.channels >= MIN_CHANNELS
      && conf.sample_rate <= MAX_SAMPLE_RATE
      && conf.sample_rate >= MIN_SAMPLE_RATE
      && sample_format == SUPPORTED_SAMPLE_FORMAT
433// Convert the given duration in frames at the given sample rate to a `std::time::Duration`.
434fn frames_to_duration(frames: usize, rate: usize) -> std::time::Duration {
  let secsf = frames as f64 / rate as f64;
  let secs = secsf as u64;
  let nanos = ((secsf - secs as f64) * 1_000_000_000.0) as u32;
  std::time::Duration::new(secs, nanos)

Lift freeq's AV media plane out of sleek 90f8b89 nandi 19d ago	1	//! Emscripten backend implementation.
	2	//!
	3	//! Default backend on Emscripten.
	4
	5	use js_sys::Float32Array;
	6	use std::panic::AssertUnwindSafe;
	7	use std::time::Duration;
	8	use wasm_bindgen::prelude::*;
	9	use wasm_bindgen::JsCast;
	10	use wasm_bindgen_futures::{spawn_local, JsFuture};
	11	use web_sys::AudioContext;
	12
	13	use crate::traits::{DeviceTrait, HostTrait, StreamTrait};
	14	use crate::{
	15	BufferSize, BuildStreamError, Data, DefaultStreamConfigError, DeviceDescription,
	16	DeviceDescriptionBuilder, DeviceId, DeviceIdError, DeviceNameError, DevicesError,
	17	InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleFormat,
	18	SampleRate, StreamConfig, StreamError, SupportedBufferSize, SupportedStreamConfig,
	19	SupportedStreamConfigRange, SupportedStreamConfigsError,
	20	};
	21
	22	// The emscripten backend currently works by instantiating an `AudioContext` object per `Stream`.
	23	// Creating a stream creates a new `AudioContext`. Destroying a stream destroys it. Creation of a
	24	// `Host` instance initializes the `stdweb` context.
	25
	26	/// The default emscripten host type.
	27	#[derive(Debug)]
	28	pub struct Host;
	29
	30	/// Content is false if the iterator is empty.
	31	pub struct Devices(bool);
	32
	33	#[derive(Clone, Debug, PartialEq, Eq)]
	34	pub struct Device;
	35
	36	#[wasm_bindgen]
	37	#[derive(Clone)]
	38	pub struct Stream {
	39	// A reference to an `AudioContext` object.
	40	audio_ctxt: AudioContext,
	41	}
	42
	43	// WASM runs in a single-threaded environment, so Send and Sync are safe by design.
	44	unsafe impl Send for Stream {}
	45	unsafe impl Sync for Stream {}
	46
	47	// Compile-time assertion that Stream is Send and Sync
	48	crate::assert_stream_send!(Stream);
	49	crate::assert_stream_sync!(Stream);
	50
	51	pub use crate::iter::{SupportedInputConfigs, SupportedOutputConfigs};
	52
	53	const MIN_CHANNELS: u16 = 1;
	54	const MAX_CHANNELS: u16 = 32;
	55	const MIN_SAMPLE_RATE: SampleRate = 8_000;
	56	const MAX_SAMPLE_RATE: SampleRate = 96_000;
	57	const DEFAULT_SAMPLE_RATE: SampleRate = 44_100;
	58	const MIN_BUFFER_SIZE: u32 = 1;
	59	const MAX_BUFFER_SIZE: u32 = u32::MAX;
	60	const DEFAULT_BUFFER_SIZE: usize = 2048;
	61	const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;
	62
	63	impl Host {
	64	pub fn new() -> Result<Self, crate::HostUnavailable> {
65	Ok(Host)
66	}
67	}
68
69	impl Devices {
70	fn new() -> Result<Self, DevicesError> {
71	Ok(Self::default())
72	}
73	}
74
75	impl Device {
76	fn description(&self) -> Result<DeviceDescription, DeviceNameError> {
77	Ok(DeviceDescriptionBuilder::new("Default Device".to_string())
78	.direction(crate::DeviceDirection::Output)
79	.build())
80	}
81
82	fn id(&self) -> Result<DeviceId, DeviceIdError> {
83	Ok(DeviceId(
84	crate::platform::HostId::Emscripten,
85	"default".to_string(),
86	))
87	}
88
89	fn supported_input_configs(
90	&self,
91	) -> Result<SupportedInputConfigs, SupportedStreamConfigsError> {
92	unimplemented!();
93	}
94
95	fn supported_output_configs(
96	&self,
97	) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
98	let buffer_size = SupportedBufferSize::Range {
99	min: MIN_BUFFER_SIZE,
100	max: MAX_BUFFER_SIZE,
101	};
102	let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
103	.map(\|channels\| SupportedStreamConfigRange {
104	channels,
105	min_sample_rate: MIN_SAMPLE_RATE,
106	max_sample_rate: MAX_SAMPLE_RATE,
107	buffer_size,
108	sample_format: SUPPORTED_SAMPLE_FORMAT,
109	})
110	.collect();
111	Ok(configs.into_iter())
112	}
113
114	fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
115	unimplemented!();
116	}
117
118	fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
119	const EXPECT: &str = "expected at least one valid webaudio stream config";
120	let config = self
121	.supported_output_configs()
122	.expect(EXPECT)
123	.max_by(\|a, b\| a.cmp_default_heuristics(b))
124	.unwrap()
125	.with_sample_rate(DEFAULT_SAMPLE_RATE);
126
127	Ok(config)
128	}
129	}
130
131	impl HostTrait for Host {
132	type Devices = Devices;
133	type Device = Device;
134
135	fn is_available() -> bool {
136	// Assume this host is always available on emscripten.
137	true
138	}
139
140	fn devices(&self) -> Result<Self::Devices, DevicesError> {
141	Devices::new()
142	}
143
144	fn default_input_device(&self) -> Option<Self::Device> {
145	default_input_device()
146	}
147
148	fn default_output_device(&self) -> Option<Self::Device> {
149	default_output_device()
150	}
151	}
152
153	impl DeviceTrait for Device {
154	type SupportedInputConfigs = SupportedInputConfigs;
155	type SupportedOutputConfigs = SupportedOutputConfigs;
156	type Stream = Stream;
157
158	fn description(&self) -> Result<DeviceDescription, DeviceNameError> {
159	Device::description(self)
160	}
161
162	fn id(&self) -> Result<DeviceId, DeviceIdError> {
163	Device::id(self)
164	}
165
166	fn supported_input_configs(
167	&self,
168	) -> Result<Self::SupportedInputConfigs, SupportedStreamConfigsError> {
169	Device::supported_input_configs(self)
170	}
171
172	fn supported_output_configs(
173	&self,
174	) -> Result<Self::SupportedOutputConfigs, SupportedStreamConfigsError> {
175	Device::supported_output_configs(self)
176	}
177
178	fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
179	Device::default_input_config(self)
180	}
181
182	fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
183	Device::default_output_config(self)
184	}
185
186	fn build_input_stream_raw<D, E>(
187	&self,
188	_config: StreamConfig,
189	_sample_format: SampleFormat,
190	_data_callback: D,
191	_error_callback: E,
192	_timeout: Option<Duration>,
193	) -> Result<Self::Stream, BuildStreamError>
194	where
195	D: FnMut(&Data, &InputCallbackInfo) + Send + 'static,
196	E: FnMut(StreamError) + Send + 'static,
197	{
198	unimplemented!()
199	}
200
201	fn build_output_stream_raw<D, E>(
202	&self,
203	config: StreamConfig,
204	sample_format: SampleFormat,
205	data_callback: D,
206	_error_callback: E,
207	_timeout: Option<Duration>,
208	) -> Result<Self::Stream, BuildStreamError>
209	where
210	D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
211	E: FnMut(StreamError) + Send + 'static,
212	{
213	if !valid_config(config, sample_format) {
214	return Err(BuildStreamError::StreamConfigNotSupported);
215	}
216
217	let buffer_size_frames = match config.buffer_size {
218	BufferSize::Fixed(v) => {
219	if !(MIN_BUFFER_SIZE..=MAX_BUFFER_SIZE).contains(&v) {
220	return Err(BuildStreamError::StreamConfigNotSupported);
221	}
222	v as usize
223	}
224	BufferSize::Default => DEFAULT_BUFFER_SIZE,
225	};
226
227	// Create the stream.
228	let audio_ctxt = AudioContext::new().expect("webaudio is not present on this system");
229	let stream = Stream { audio_ctxt };
230
231	// Use `set_timeout` to invoke a Rust callback repeatedly.
232	//
233	// The job of this callback is to fill the content of the audio buffers.
234	//
235	// See also: The call to `set_timeout` at the end of the `audio_callback_fn` which creates
236	// the loop.
237	let data_callback = AssertUnwindSafe(data_callback);
238	set_timeout(
239	10,
240	stream.clone(),
241	data_callback,
242	config,
243	sample_format,
244	buffer_size_frames as u32,
245	);
246
247	Ok(stream)
248	}
249	}
250
251	impl StreamTrait for Stream {
252	fn play(&self) -> Result<(), PlayStreamError> {
253	let future = JsFuture::from(
254	self.audio_ctxt
255	.resume()
256	.expect("Could not resume the stream"),
257	);
258	spawn_local(async {
259	match future.await {
260	Ok(value) => assert!(value.is_undefined()),
261	Err(value) => panic!("AudioContext.resume() promise was rejected: {:?}", value),
262	}
263	});
264	Ok(())
265	}
266
267	fn pause(&self) -> Result<(), PauseStreamError> {
268	let future = JsFuture::from(
269	self.audio_ctxt
270	.suspend()
271	.expect("Could not suspend the stream"),
272	);
273	spawn_local(async {
274	match future.await {
275	Ok(value) => assert!(value.is_undefined()),
276	Err(value) => panic!("AudioContext.suspend() promise was rejected: {:?}", value),
277	}
278	});
279	Ok(())
280	}
281	}
282
283	fn audio_callback_fn<D>(
284	mut data_callback: AssertUnwindSafe<D>,
285	) -> impl FnOnce(Stream, StreamConfig, SampleFormat, u32)
286	where
287	D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
288	{
289	\|stream, config, sample_format, buffer_size_frames\| {
290	let sample_rate = config.sample_rate;
291	let buffer_size_samples = buffer_size_frames * config.channels as u32;
292	let audio_ctxt = &stream.audio_ctxt;
293
294	// TODO: We should be re-using a buffer.
295	let mut temporary_buffer = vec![0f32; buffer_size_samples as usize];
296
297	{
298	let len = temporary_buffer.len();
299	let data = temporary_buffer.as_mut_ptr() as *mut ();
300	let mut data = unsafe { Data::from_parts(data, len, sample_format) };
301	let now_secs: f64 = audio_ctxt.current_time();
302	let callback = crate::StreamInstant::from_secs_f64(now_secs);
303	// TODO: Use proper latency instead. Currently, unsupported on most browsers though, so
304	// we estimate based on buffer size instead. Probably should use this, but it's only
305	// supported by firefox (2020-04-28).
306	// let latency_secs: f64 = audio_ctxt.outputLatency.try_into().unwrap();
307	let buffer_duration = frames_to_duration(len, sample_rate as usize);
308	let playback = callback
309	.add(buffer_duration)
310	.expect("`playback` occurs beyond representation supported by `StreamInstant`");
311	let timestamp = crate::OutputStreamTimestamp { callback, playback };
312	let info = OutputCallbackInfo { timestamp };
313	data_callback(&mut data, &info);
314	}
315
316	let typed_array: Float32Array = temporary_buffer.as_slice().into();
317
318	debug_assert_eq!(temporary_buffer.len() % config.channels as usize, 0);
319
320	let src_buffer = Float32Array::new(typed_array.buffer().as_ref());
321	let context = audio_ctxt;
322	let buffer = context
323	.create_buffer(
324	config.channels as u32,
325	buffer_size_frames,
326	sample_rate as f32,
327	)
328	.expect("Buffer could not be created");
329	for channel in 0..config.channels {
330	let mut buffer_content = buffer
331	.get_channel_data(channel as u32)
332	.expect("Should be impossible");
333	for (i, buffer_content_item) in buffer_content.iter_mut().enumerate() {
334	*buffer_content_item =
335	src_buffer.get_index(i as u32 * config.channels as u32 + channel as u32);
336	}
337	}
338
339	let node = context
340	.create_buffer_source()
341	.expect("The buffer source node could not be created");
342	node.set_buffer(Some(&buffer));
343	context
344	.destination()
345	.connect_with_audio_node(&node)
346	.expect("Could not connect the audio node to the destination");
347	node.start().expect("Could not start the audio node");
348
349	// TODO: handle latency better ; right now we just use setInterval with the amount of sound
350	// data that is in each buffer ; this is obviously bad, and also the schedule is too tight
351	// and there may be underflows
352	set_timeout(
353	1000 * buffer_size_frames as i32 / sample_rate as i32,
354	stream.clone().clone(),
355	data_callback,
356	config,
357	sample_format,
358	buffer_size_frames,
359	);
360	}
361	}
362
363	fn set_timeout<D>(
364	time: i32,
365	stream: Stream,
366	data_callback: AssertUnwindSafe<D>,
367	config: StreamConfig,
368	sample_format: SampleFormat,
369	buffer_size_frames: u32,
370	) where
371	D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
372	{
373	let window = web_sys::window().expect("Not in a window somehow?");
374	window
375	.set_timeout_with_callback_and_timeout_and_arguments_4(
376	Closure::once_into_js(audio_callback_fn(data_callback))
377	.dyn_ref::<js_sys::Function>()
378	.expect("The function was somehow not a function"),
379	time,
380	&stream.into(),
381	&config.into(),
382	&Closure::once_into_js(move \|\| sample_format),
383	&buffer_size_frames.into(),
384	)
385	.expect("The timeout could not be set");
386	}
387
388	impl Default for Devices {
389	fn default() -> Devices {
390	// We produce an empty iterator if the WebAudio API isn't available.
391	Devices(is_webaudio_available())
392	}
393	}
394	impl Iterator for Devices {
395	type Item = Device;
396
397	fn next(&mut self) -> Option<Device> {
398	if self.0 {
399	self.0 = false;
400	Some(Device)
401	} else {
402	None
403	}
404	}
405	}
406
407	fn default_input_device() -> Option<Device> {
408	unimplemented!();
409	}
410
411	fn default_output_device() -> Option<Device> {
412	if is_webaudio_available() {
413	Some(Device)
414	} else {
415	None
416	}
417	}
418
419	// Detects whether the `AudioContext` global variable is available.
420	fn is_webaudio_available() -> bool {
421	AudioContext::new().is_ok()
422	}
423
424	// Whether or not the given stream configuration is valid for building a stream.
425	fn valid_config(conf: StreamConfig, sample_format: SampleFormat) -> bool {
426	conf.channels <= MAX_CHANNELS
427	&& conf.channels >= MIN_CHANNELS
428	&& conf.sample_rate <= MAX_SAMPLE_RATE
429	&& conf.sample_rate >= MIN_SAMPLE_RATE
430	&& sample_format == SUPPORTED_SAMPLE_FORMAT
431	}
432
433	// Convert the given duration in frames at the given sample rate to a `std::time::Duration`.
434	fn frames_to_duration(frames: usize, rate: usize) -> std::time::Duration {
435	let secsf = frames as f64 / rate as f64;
436	let secs = secsf as u64;
437	let nanos = ((secsf - secs as f64) * 1_000_000_000.0) as u32;
438	std::time::Duration::new(secs, nanos)
439	}