Add implementation of supported stream configs for webaudio
The `supported_stream_configs` method now returns the range of configurations that are required to be supported for `BaseAudioContext.createBuffer()` as mentioned here: https://developer.mozilla.org/en-US/docs/Web/API/BaseAudioContext/createBuffer That is, valid stream configurations are now considered to be any configuration that has: - 1 <= channel_count <= 32 and - 8khz <= sample_rate <= 96khz - sample_format == f32 Closes #410. Closes #411.
This commit is contained in:
mitchmindtree
parent
cf4e6ca5bf
commit
1dfdeace25
|
|
@ -7,15 +7,14 @@ use self::wasm_bindgen::prelude::*;
|
|||
use self::wasm_bindgen::JsCast;
|
||||
use self::web_sys::{AudioContext, AudioContextOptions};
|
||||
use crate::{
|
||||
BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
|
||||
InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleRate,
|
||||
StreamConfig, StreamError, SupportedStreamConfig, SupportedStreamConfigRange,
|
||||
SupportedStreamConfigsError,
|
||||
BackendSpecificError, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError,
|
||||
DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError,
|
||||
SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
|
||||
SupportedStreamConfigRange, SupportedStreamConfigsError,
|
||||
};
|
||||
use std::ops::DerefMut;
|
||||
use std::sync::{Arc, Mutex, RwLock};
|
||||
use traits::{DeviceTrait, HostTrait, StreamTrait};
|
||||
use {BackendSpecificError, SampleFormat};
|
||||
|
||||
/// Content is false if the iterator is empty.
|
||||
pub struct Devices(bool);
|
||||
|
|
@ -28,11 +27,20 @@ pub struct Host;
|
|||
pub struct Stream {
|
||||
ctx: Arc<AudioContext>,
|
||||
on_ended_closures: Vec<Arc<RwLock<Option<Closure<dyn FnMut()>>>>>,
|
||||
config: StreamConfig,
|
||||
buffer_size_frames: usize,
|
||||
}
|
||||
|
||||
pub type SupportedInputConfigs = ::std::vec::IntoIter<SupportedStreamConfigRange>;
|
||||
pub type SupportedOutputConfigs = ::std::vec::IntoIter<SupportedStreamConfigRange>;
|
||||
|
||||
const MIN_CHANNELS: u16 = 1;
|
||||
const MAX_CHANNELS: u16 = 32;
|
||||
const MIN_SAMPLE_RATE: SampleRate = SampleRate(8_000);
|
||||
const MAX_SAMPLE_RATE: SampleRate = SampleRate(96_000);
|
||||
const DEFAULT_SAMPLE_RATE: SampleRate = SampleRate(44_100);
|
||||
const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;
|
||||
|
||||
impl Host {
|
||||
pub fn new() -> Result<Self, crate::HostUnavailable> {
|
||||
Ok(Host)
|
||||
|
|
@ -84,21 +92,15 @@ impl Device {
|
|||
fn supported_output_configs(
|
||||
&self,
|
||||
) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
|
||||
// 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![SupportedStreamConfigRange {
|
||||
channels: 2,
|
||||
min_sample_rate: SampleRate(44100),
|
||||
max_sample_rate: SampleRate(44100),
|
||||
sample_format: ::SampleFormat::F32,
|
||||
}]
|
||||
.into_iter())
|
||||
let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
|
||||
.map(|channels| SupportedStreamConfigRange {
|
||||
channels,
|
||||
min_sample_rate: MIN_SAMPLE_RATE,
|
||||
max_sample_rate: MAX_SAMPLE_RATE,
|
||||
sample_format: SUPPORTED_SAMPLE_FORMAT,
|
||||
})
|
||||
.collect();
|
||||
Ok(configs.into_iter())
|
||||
}
|
||||
|
||||
#[inline]
|
||||
|
|
@ -108,12 +110,15 @@ impl Device {
|
|||
|
||||
#[inline]
|
||||
fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
|
||||
// TODO: because it is hard coded, see supported_output_formats.
|
||||
Ok(SupportedStreamConfig {
|
||||
channels: 2,
|
||||
sample_rate: ::SampleRate(44100),
|
||||
sample_format: ::SampleFormat::F32,
|
||||
})
|
||||
const EXPECT: &str = "expected at least one valid webaudio stream config";
|
||||
let mut configs: Vec<_> = self.supported_output_configs().expect(EXPECT).collect();
|
||||
configs.sort_by(|a, b| a.cmp_default_heuristics(b));
|
||||
let config = configs
|
||||
.into_iter()
|
||||
.next()
|
||||
.expect(EXPECT)
|
||||
.with_sample_rate(DEFAULT_SAMPLE_RATE);
|
||||
Ok(config)
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -177,14 +182,16 @@ impl DeviceTrait for Device {
|
|||
D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
|
||||
E: FnMut(StreamError) + Send + 'static,
|
||||
{
|
||||
assert_eq!(
|
||||
sample_format,
|
||||
SampleFormat::F32,
|
||||
"WebAudio backend currently only supports `f32` data",
|
||||
);
|
||||
if !valid_config(config, sample_format) {
|
||||
return Err(BuildStreamError::StreamConfigNotSupported);
|
||||
}
|
||||
|
||||
let n_channels = config.channels as usize;
|
||||
// Use a buffer period of 1/3s for this early proof of concept.
|
||||
let buffer_length = (config.sample_rate.0 as f64 / 3.0).round() as usize;
|
||||
// TODO: Change this to the requested buffer size when updating for the buffer size API.
|
||||
let buffer_size_frames = (config.sample_rate.0 as f64 / 3.0).round() as usize;
|
||||
let buffer_size_samples = buffer_size_frames * n_channels;
|
||||
let buffer_time_step_secs = buffer_time_step_secs(buffer_size_frames, config.sample_rate);
|
||||
let data_callback = Arc::new(Mutex::new(Box::new(data_callback)));
|
||||
|
||||
// Create the WebAudio stream.
|
||||
|
|
@ -206,22 +213,23 @@ impl DeviceTrait for Device {
|
|||
// A cursor keeping track of the current time at which new frames should be scheduled.
|
||||
let time = Arc::new(RwLock::new(0f64));
|
||||
|
||||
// Create a set of closures / callbacks which will continuously fetch and schedule sample playback.
|
||||
// Starting with two workers, eg a front and back buffer so that audio frames can be fetched in the background.
|
||||
// Create a set of closures / callbacks which will continuously fetch and schedule sample
|
||||
// playback. Starting with two workers, eg a front and back buffer so that audio frames
|
||||
// can be fetched in the background.
|
||||
for _i in 0..2 {
|
||||
let data_callback_handle = data_callback.clone();
|
||||
let ctx_handle = ctx.clone();
|
||||
let time_handle = time.clone();
|
||||
|
||||
// A set of temporary buffers to be used for intermediate sample transformation steps.
|
||||
let mut temporary_buffer = vec![0f32; buffer_length * config.channels as usize];
|
||||
let mut temporary_channel_buffer = vec![0f32; buffer_length];
|
||||
let mut temporary_buffer = vec![0f32; buffer_size_samples];
|
||||
let mut temporary_channel_buffer = vec![0f32; buffer_size_frames];
|
||||
|
||||
// Create a webaudio buffer which will be reused to avoid allocations.
|
||||
let ctx_buffer = ctx
|
||||
.create_buffer(
|
||||
config.channels as u32,
|
||||
buffer_length as u32,
|
||||
buffer_size_frames as u32,
|
||||
config.sample_rate.0 as f32,
|
||||
)
|
||||
.map_err(|err| -> BuildStreamError {
|
||||
|
|
@ -235,9 +243,6 @@ impl DeviceTrait for Device {
|
|||
Arc::new(RwLock::new(None));
|
||||
let on_ended_closure_handle = on_ended_closure.clone();
|
||||
|
||||
let n_channels = config.channels as usize;
|
||||
let sample_rate = config.sample_rate.0 as f64;
|
||||
|
||||
on_ended_closure
|
||||
.write()
|
||||
.unwrap()
|
||||
|
|
@ -247,11 +252,13 @@ impl DeviceTrait for Device {
|
|||
let time_at_start_of_buffer = time_handle
|
||||
.read()
|
||||
.expect("Unable to get a read lock on the time cursor");
|
||||
// Synchronise first buffer as necessary (eg. keep the time value referenced to the context clock).
|
||||
// Synchronise first buffer as necessary (eg. keep the time value
|
||||
// referenced to the context clock).
|
||||
if *time_at_start_of_buffer > 0.001 {
|
||||
*time_at_start_of_buffer
|
||||
} else {
|
||||
// 25ms of time to fetch the first sample data, increase to avoid initial underruns.
|
||||
// 25ms of time to fetch the first sample data, increase to avoid
|
||||
// initial underruns.
|
||||
now + 0.025
|
||||
}
|
||||
};
|
||||
|
|
@ -260,7 +267,6 @@ impl DeviceTrait for Device {
|
|||
{
|
||||
let len = temporary_buffer.len();
|
||||
let data = temporary_buffer.as_mut_ptr() as *mut ();
|
||||
let sample_format = SampleFormat::F32;
|
||||
let mut data = unsafe { Data::from_parts(data, len, sample_format) };
|
||||
let mut data_callback = data_callback_handle.lock().unwrap();
|
||||
let callback = crate::StreamInstant::from_secs_f64(now);
|
||||
|
|
@ -274,7 +280,7 @@ impl DeviceTrait for Device {
|
|||
// We do not reference the audio context buffer directly eg getChannelData.
|
||||
// As wasm-bindgen only gives us a copy, not a direct reference.
|
||||
for channel in 0..n_channels {
|
||||
for i in 0..buffer_length {
|
||||
for i in 0..buffer_size_frames {
|
||||
temporary_channel_buffer[i] =
|
||||
temporary_buffer[n_channels * i + channel];
|
||||
}
|
||||
|
|
@ -283,7 +289,8 @@ impl DeviceTrait for Device {
|
|||
.expect("Unable to write sample data into the audio context buffer");
|
||||
}
|
||||
|
||||
// Create an AudioBufferSourceNode, scheduled it to playback the reused buffer in the future.
|
||||
// Create an AudioBufferSourceNode, schedule it to playback the reused buffer
|
||||
// in the future.
|
||||
let source = ctx_handle
|
||||
.create_buffer_source()
|
||||
.expect("Unable to create a webaudio buffer source");
|
||||
|
|
@ -308,8 +315,7 @@ impl DeviceTrait for Device {
|
|||
.expect("Unable to start the webaudio buffer source");
|
||||
|
||||
// Keep track of when the next buffer worth of samples should be played.
|
||||
*time_handle.write().unwrap() =
|
||||
time_at_start_of_buffer + (buffer_length as f64 / sample_rate);
|
||||
*time_handle.write().unwrap() = time_at_start_of_buffer + buffer_time_step_secs;
|
||||
}) as Box<dyn FnMut()>));
|
||||
|
||||
on_ended_closures.push(on_ended_closure);
|
||||
|
|
@ -318,6 +324,8 @@ impl DeviceTrait for Device {
|
|||
Ok(Stream {
|
||||
ctx,
|
||||
on_ended_closures,
|
||||
config: config.clone(),
|
||||
buffer_size_frames,
|
||||
})
|
||||
}
|
||||
}
|
||||
|
|
@ -327,8 +335,12 @@ impl StreamTrait for Stream {
|
|||
let window = web_sys::window().unwrap();
|
||||
match self.ctx.resume() {
|
||||
Ok(_) => {
|
||||
// Begin webaudio playback, initially scheduling the closures to fire on a timeout event.
|
||||
// Begin webaudio playback, initially scheduling the closures to fire on a timeout
|
||||
// event.
|
||||
let mut offset_ms = 10;
|
||||
let time_step_secs =
|
||||
buffer_time_step_secs(self.buffer_size_frames, self.config.sample_rate);
|
||||
let time_step_ms = (time_step_secs * 1_000.0) as i32;
|
||||
for on_ended_closure in self.on_ended_closures.iter() {
|
||||
window
|
||||
.set_timeout_with_callback_and_timeout_and_arguments_0(
|
||||
|
|
@ -342,7 +354,7 @@ impl StreamTrait for Stream {
|
|||
offset_ms,
|
||||
)
|
||||
.unwrap();
|
||||
offset_ms += 333 / 2;
|
||||
offset_ms += time_step_ms;
|
||||
}
|
||||
Ok(())
|
||||
}
|
||||
|
|
@ -414,3 +426,16 @@ fn is_webaudio_available() -> bool {
|
|||
false
|
||||
}
|
||||
}
|
||||
|
||||
// Whether or not the given stream configuration is valid for building a stream.
|
||||
fn 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
|
||||
}
|
||||
|
||||
fn buffer_time_step_secs(buffer_size_frames: usize, sample_rate: SampleRate) -> f64 {
|
||||
buffer_size_frames as f64 / sample_rate.0 as f64
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue