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Merge pull request #415 from mitchmindtree/webaudio_config 

Add implementation of supported stream configs for webaudio
This commit is contained in:
mitchmindtree 2020-05-26 14:53:37 +02:00 committed by GitHub
parent 713eddd89a f03fd69b65
commit 5fa5ce593d
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GPG Key ID: 4AEE18F83AFDEB23
1 changed files with 82 additions and 53 deletions
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135
src/host/webaudio/mod.rs
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@ -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)
@ -77,43 +85,42 @@ impl Device {
fn supported_input_configs(
&self,
) -> Result<SupportedInputConfigs, SupportedStreamConfigsError> {
unimplemented!();
// TODO
Ok(Vec::new().into_iter())
}
#[inline]
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]
fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
unimplemented!();
// TODO
Err(DefaultStreamConfigError::StreamTypeNotSupported)
}
#[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)
}
}
@ -162,7 +169,8 @@ impl DeviceTrait for Device {
D: FnMut(&Data, &InputCallbackInfo) + Send + 'static,
E: FnMut(StreamError) + Send + 'static,
{
unimplemented!()
// TODO
Err(BuildStreamError::StreamConfigNotSupported)
}
/// Create an output stream.
@ -177,14 +185,16 @@ 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 +216,23 @@ 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 +246,6 @@ 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 +255,13 @@ 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 +270,6 @@ 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 +283,7 @@ 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 +292,8 @@ 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 +318,7 @@ 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 +327,8 @@ impl DeviceTrait for Device {
Ok(Stream {
ctx,
on_ended_closures,
config: config.clone(),
buffer_size_frames,
})
}
}
@ -327,8 +338,12 @@ 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 +357,7 @@ impl StreamTrait for Stream {
offset_ms,
)
.unwrap();
offset_ms += 333 / 2;
offset_ms += time_step_ms;
}
Ok(())
}
@ -394,7 +409,8 @@ impl Iterator for Devices {
#[inline]
fn default_input_device() -> Option<Device> {
unimplemented!();
// TODO
None
}
#[inline]
@ -414,3 +430,16 @@ 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
}