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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 2020-05-25 13:19:52 +02:00
parent cf4e6ca5bf
commit 1dfdeace25
1 changed files with 74 additions and 49 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

123
src/host/webaudio/mod.rs
View File

@ -7,15 +7,14 @@ use self::wasm_bindgen::prelude::*;
use self::wasm_bindgen::JsCast; use self::wasm_bindgen::JsCast;
use self::web_sys::{AudioContext, AudioContextOptions}; use self::web_sys::{AudioContext, AudioContextOptions};
use crate::{ use crate::{
BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError, BackendSpecificError, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError,
InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleRate, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError,
StreamConfig, StreamError, SupportedStreamConfig, SupportedStreamConfigRange, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
SupportedStreamConfigsError, SupportedStreamConfigRange, SupportedStreamConfigsError,
}; };
use std::ops::DerefMut; use std::ops::DerefMut;
use std::sync::{Arc, Mutex, RwLock}; use std::sync::{Arc, Mutex, RwLock};
use traits::{DeviceTrait, HostTrait, StreamTrait}; use traits::{DeviceTrait, HostTrait, StreamTrait};
use {BackendSpecificError, SampleFormat};
/// Content is false if the iterator is empty. /// Content is false if the iterator is empty.
pub struct Devices(bool); pub struct Devices(bool);
@ -28,11 +27,20 @@ pub struct Host;
pub struct Stream { pub struct Stream {
ctx: Arc<AudioContext>, ctx: Arc<AudioContext>,
on_ended_closures: Vec<Arc<RwLock<Option<Closure<dyn FnMut()>>>>>, 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 SupportedInputConfigs = ::std::vec::IntoIter<SupportedStreamConfigRange>;
pub type SupportedOutputConfigs = ::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 { impl Host {
pub fn new() -> Result<Self, crate::HostUnavailable> { pub fn new() -> Result<Self, crate::HostUnavailable> {
Ok(Host) Ok(Host)
@ -84,21 +92,15 @@ impl Device {
fn supported_output_configs( fn supported_output_configs(
&self, &self,
) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> { ) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
// TODO: right now cpal's API doesn't allow flexibility here let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
// "44100" and "2" (channels) have also been hard-coded in the rest of the code ; if .map(|channels| SupportedStreamConfigRange {
// this ever becomes more flexible, don't forget to change that channels,
// According to https://developer.mozilla.org/en-US/docs/Web/API/BaseAudioContext/createBuffer min_sample_rate: MIN_SAMPLE_RATE,
// browsers must support 1 to 32 channels at leats and 8,000 Hz to 96,000 Hz. max_sample_rate: MAX_SAMPLE_RATE,
// sample_format: SUPPORTED_SAMPLE_FORMAT,
// 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. .collect();
Ok(vec![SupportedStreamConfigRange { Ok(configs.into_iter())
channels: 2,
min_sample_rate: SampleRate(44100),
max_sample_rate: SampleRate(44100),
sample_format: ::SampleFormat::F32,
}]
.into_iter())
} }
#[inline] #[inline]
@ -108,12 +110,15 @@ impl Device {
#[inline] #[inline]
fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> { fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
// TODO: because it is hard coded, see supported_output_formats. const EXPECT: &str = "expected at least one valid webaudio stream config";
Ok(SupportedStreamConfig { let mut configs: Vec<_> = self.supported_output_configs().expect(EXPECT).collect();
channels: 2, configs.sort_by(|a, b| a.cmp_default_heuristics(b));
sample_rate: ::SampleRate(44100), let config = configs
sample_format: ::SampleFormat::F32, .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, D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
E: FnMut(StreamError) + Send + 'static, E: FnMut(StreamError) + Send + 'static,
{ {
assert_eq!( if !valid_config(config, sample_format) {
sample_format, return Err(BuildStreamError::StreamConfigNotSupported);
SampleFormat::F32, }
"WebAudio backend currently only supports `f32` data",
);
let n_channels = config.channels as usize;
// Use a buffer period of 1/3s for this early proof of concept. // 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))); let data_callback = Arc::new(Mutex::new(Box::new(data_callback)));
// Create the WebAudio stream. // 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. // A cursor keeping track of the current time at which new frames should be scheduled.
let time = Arc::new(RwLock::new(0f64)); let time = Arc::new(RwLock::new(0f64));
// Create a set of closures / callbacks which will continuously fetch and schedule sample playback. // Create a set of closures / callbacks which will continuously fetch and schedule sample
// Starting with two workers, eg a front and back buffer so that audio frames can be fetched in the background. // 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 { for _i in 0..2 {
let data_callback_handle = data_callback.clone(); let data_callback_handle = data_callback.clone();
let ctx_handle = ctx.clone(); let ctx_handle = ctx.clone();
let time_handle = time.clone(); let time_handle = time.clone();
// A set of temporary buffers to be used for intermediate sample transformation steps. // 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_buffer = vec![0f32; buffer_size_samples];
let mut temporary_channel_buffer = vec![0f32; buffer_length]; let mut temporary_channel_buffer = vec![0f32; buffer_size_frames];
// Create a webaudio buffer which will be reused to avoid allocations. // Create a webaudio buffer which will be reused to avoid allocations.
let ctx_buffer = ctx let ctx_buffer = ctx
.create_buffer( .create_buffer(
config.channels as u32, config.channels as u32,
buffer_length as u32, buffer_size_frames as u32,
config.sample_rate.0 as f32, config.sample_rate.0 as f32,
) )
.map_err(|err| -> BuildStreamError { .map_err(|err| -> BuildStreamError {
@ -235,9 +243,6 @@ impl DeviceTrait for Device {
Arc::new(RwLock::new(None)); Arc::new(RwLock::new(None));
let on_ended_closure_handle = on_ended_closure.clone(); 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 on_ended_closure
.write() .write()
.unwrap() .unwrap()
@ -247,11 +252,13 @@ impl DeviceTrait for Device {
let time_at_start_of_buffer = time_handle let time_at_start_of_buffer = time_handle
.read() .read()
.expect("Unable to get a read lock on the time cursor"); .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 { if *time_at_start_of_buffer > 0.001 {
*time_at_start_of_buffer *time_at_start_of_buffer
} else { } 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 now + 0.025
} }
}; };
@ -260,7 +267,6 @@ impl DeviceTrait for Device {
{ {
let len = temporary_buffer.len(); let len = temporary_buffer.len();
let data = temporary_buffer.as_mut_ptr() as *mut (); 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 = unsafe { Data::from_parts(data, len, sample_format) };
let mut data_callback = data_callback_handle.lock().unwrap(); let mut data_callback = data_callback_handle.lock().unwrap();
let callback = crate::StreamInstant::from_secs_f64(now); 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. // We do not reference the audio context buffer directly eg getChannelData.
// As wasm-bindgen only gives us a copy, not a direct reference. // As wasm-bindgen only gives us a copy, not a direct reference.
for channel in 0..n_channels { for channel in 0..n_channels {
for i in 0..buffer_length { for i in 0..buffer_size_frames {
temporary_channel_buffer[i] = temporary_channel_buffer[i] =
temporary_buffer[n_channels * i + channel]; 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"); .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 let source = ctx_handle
.create_buffer_source() .create_buffer_source()
.expect("Unable to create a webaudio 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"); .expect("Unable to start the webaudio buffer source");
// Keep track of when the next buffer worth of samples should be played. // Keep track of when the next buffer worth of samples should be played.
*time_handle.write().unwrap() = *time_handle.write().unwrap() = time_at_start_of_buffer + buffer_time_step_secs;
time_at_start_of_buffer + (buffer_length as f64 / sample_rate);
}) as Box<dyn FnMut()>)); }) as Box<dyn FnMut()>));
on_ended_closures.push(on_ended_closure); on_ended_closures.push(on_ended_closure);
@ -318,6 +324,8 @@ impl DeviceTrait for Device {
Ok(Stream { Ok(Stream {
ctx, ctx,
on_ended_closures, on_ended_closures,
config: config.clone(),
buffer_size_frames,
}) })
} }
} }
@ -327,8 +335,12 @@ impl StreamTrait for Stream {
let window = web_sys::window().unwrap(); let window = web_sys::window().unwrap();
match self.ctx.resume() { match self.ctx.resume() {
Ok(_) => { 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 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() { for on_ended_closure in self.on_ended_closures.iter() {
window window
.set_timeout_with_callback_and_timeout_and_arguments_0( .set_timeout_with_callback_and_timeout_and_arguments_0(
@ -342,7 +354,7 @@ impl StreamTrait for Stream {
offset_ms, offset_ms,
) )
.unwrap(); .unwrap();
offset_ms += 333 / 2; offset_ms += time_step_ms;
} }
Ok(()) Ok(())
} }
@ -414,3 +426,16 @@ fn is_webaudio_available() -> bool {
false 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
}