Merge pull request #406 from mitchmindtree/webaudio-poc-rebased

Rebase/Update webaudio PR for recent breaking changes

.github/workflows/cpal.yml

@@ -108,6 +108,25 @@ jobs:
     - name: Build beep example
       run: cargo build --example beep --target ${{ matrix.target }}
 
+  wasm32-bindgen-test:
+
+    strategy:
+      matrix:
+        target: [wasm32-unknown-unknown]
+
+    runs-on: ubuntu-latest
+
+    steps:
+      - uses: actions/checkout@v1
+      - name: Install stable
+        uses: actions-rs/toolchain@v1
+        with:
+          profile: minimal
+          toolchain: stable
+          target: ${{ matrix.target }}
+      - name: Build beep example
+        run: cargo build --example beep --target ${{ matrix.target }} --features=wasm-bindgen
+
   windows-test:
     strategy:
       matrix:

Cargo.toml

@@ -38,3 +38,8 @@ mach = "0.3" # For access to mach_timebase type.
 
 [target.'cfg(target_os = "emscripten")'.dependencies]
 stdweb = { version = "0.1.3", default-features = false }
+
+[target.'cfg(all(target_arch = "wasm32", target_os = "unknown"))'.dependencies]
+wasm-bindgen = { version = "0.2.58", optional = true }
+js-sys = { version = "0.3.35" }
+web-sys = { version = "0.3.35", features = [ "AudioContext", "AudioContextOptions", "AudioBuffer", "AudioBufferSourceNode", "AudioNode",  "AudioDestinationNode", "Window", "AudioContextState"] }

src/host/emscripten/mod.rs

@@ -248,11 +248,7 @@ where
             let now_secs: f64 = js!(@{audio_ctxt}.getOutputTimestamp().currentTime)
                 .try_into()
                 .expect("failed to retrieve Value as f64");
-            let callback = {
-                let secs = now_secs as i64;
-                let nanos = ((now_secs * 1_000_000_000.0) - secs as f64 * 1_000_000_000.0) as u32;
-                crate::StreamInstant::new(secs, nanos)
-            };
+            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).

src/host/mod.rs

@@ -9,3 +9,5 @@ pub(crate) mod emscripten;
 pub(crate) mod null;
 #[cfg(windows)]
 pub(crate) mod wasapi;
+#[cfg(all(target_arch = "wasm32", feature = "wasm-bindgen"))]
+pub(crate) mod webaudio;

src/host/webaudio/mod.rs

@@ -0,0 +1,416 @@
+extern crate js_sys;
+extern crate wasm_bindgen;
+extern crate web_sys;
+
+use self::js_sys::eval;
+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,
+};
+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);
+
+#[derive(Clone, Debug, PartialEq, Eq)]
+pub struct Device;
+
+pub struct Host;
+
+pub struct Stream {
+    ctx: Arc<AudioContext>,
+    on_ended_closures: Vec<Arc<RwLock<Option<Closure<dyn FnMut()>>>>>,
+}
+
+pub type SupportedInputConfigs = ::std::vec::IntoIter<SupportedStreamConfigRange>;
+pub type SupportedOutputConfigs = ::std::vec::IntoIter<SupportedStreamConfigRange>;
+
+impl Host {
+    pub fn new() -> Result<Self, crate::HostUnavailable> {
+        Ok(Host)
+    }
+}
+
+impl HostTrait for Host {
+    type Devices = Devices;
+    type Device = Device;
+
+    fn is_available() -> bool {
+        // Assume this host is always available on webaudio.
+        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()
+    }
+}
+
+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_configs(
+        &self,
+    ) -> Result<SupportedInputConfigs, SupportedStreamConfigsError> {
+        unimplemented!();
+    }
+
+    #[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())
+    }
+
+    #[inline]
+    fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
+        unimplemented!();
+    }
+
+    #[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,
+        })
+    }
+}
+
+impl DeviceTrait for Device {
+    type SupportedInputConfigs = SupportedInputConfigs;
+    type SupportedOutputConfigs = SupportedOutputConfigs;
+    type Stream = Stream;
+
+    #[inline]
+    fn name(&self) -> Result<String, DeviceNameError> {
+        Device::name(self)
+    }
+
+    #[inline]
+    fn supported_input_configs(
+        &self,
+    ) -> Result<Self::SupportedInputConfigs, SupportedStreamConfigsError> {
+        Device::supported_input_configs(self)
+    }
+
+    #[inline]
+    fn supported_output_configs(
+        &self,
+    ) -> Result<Self::SupportedOutputConfigs, SupportedStreamConfigsError> {
+        Device::supported_output_configs(self)
+    }
+
+    #[inline]
+    fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
+        Device::default_input_config(self)
+    }
+
+    #[inline]
+    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,
+    ) -> Result<Self::Stream, BuildStreamError>
+    where
+        D: FnMut(&Data, &InputCallbackInfo) + Send + 'static,
+        E: FnMut(StreamError) + Send + 'static,
+    {
+        unimplemented!()
+    }
+
+    /// Create an output stream.
+    fn build_output_stream_raw<D, E>(
+        &self,
+        config: &StreamConfig,
+        sample_format: SampleFormat,
+        data_callback: D,
+        _error_callback: E,
+    ) -> Result<Self::Stream, BuildStreamError>
+    where
+        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",
+        );
+
+        // 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;
+        let data_callback = Arc::new(Mutex::new(Box::new(data_callback)));
+
+        // Create the WebAudio stream.
+        let mut stream_opts = AudioContextOptions::new();
+        stream_opts.sample_rate(config.sample_rate.0 as f32);
+        let ctx = Arc::new(
+            AudioContext::new_with_context_options(&stream_opts).map_err(
+                |err| -> BuildStreamError {
+                    let description = format!("{:?}", err);
+                    let err = BackendSpecificError { description };
+                    err.into()
+                },
+            )?,
+        );
+
+        // A container for managing the lifecycle of the audio callbacks.
+        let mut on_ended_closures: Vec<Arc<RwLock<Option<Closure<dyn FnMut()>>>>> = Vec::new();
+
+        // 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.
+        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];
+
+            // 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,
+                    config.sample_rate.0 as f32,
+                )
+                .map_err(|err| -> BuildStreamError {
+                    let description = format!("{:?}", err);
+                    let err = BackendSpecificError { description };
+                    err.into()
+                })?;
+
+            // A self reference to this closure for passing to future audio event calls.
+            let on_ended_closure: Arc<RwLock<Option<Closure<dyn FnMut()>>>> =
+                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()
+                .replace(Closure::wrap(Box::new(move || {
+                    let now = ctx_handle.current_time();
+                    let time_at_start_of_buffer = {
+                        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).
+                        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.
+                            now + 0.025
+                        }
+                    };
+
+                    // Populate the sample data into an interleaved temporary buffer.
+                    {
+                        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);
+                        let playback = crate::StreamInstant::from_secs_f64(time_at_start_of_buffer);
+                        let timestamp = crate::OutputStreamTimestamp { callback, playback };
+                        let info = OutputCallbackInfo { timestamp };
+                        (data_callback.deref_mut())(&mut data, &info);
+                    }
+
+                    // Deinterleave the sample data and copy into the audio context buffer.
+                    // 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 {
+                            temporary_channel_buffer[i] =
+                                temporary_buffer[n_channels * i + channel];
+                        }
+                        ctx_buffer
+                            .copy_to_channel(&mut temporary_channel_buffer, channel as i32)
+                            .expect("Unable to write sample data into the audio context buffer");
+                    }
+
+                    // Create an AudioBufferSourceNode, scheduled it to playback the reused buffer in the future.
+                    let source = ctx_handle
+                        .create_buffer_source()
+                        .expect("Unable to create a webaudio buffer source");
+                    source.set_buffer(Some(&ctx_buffer));
+                    source
+                        .connect_with_audio_node(&ctx_handle.destination())
+                        .expect(
+                        "Unable to connect the web audio buffer source to the context destination",
+                    );
+                    source.set_onended(Some(
+                        on_ended_closure_handle
+                            .read()
+                            .unwrap()
+                            .as_ref()
+                            .unwrap()
+                            .as_ref()
+                            .unchecked_ref(),
+                    ));
+
+                    source
+                        .start_with_when(time_at_start_of_buffer)
+                        .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);
+                }) as Box<dyn FnMut()>));
+
+            on_ended_closures.push(on_ended_closure);
+        }
+
+        Ok(Stream {
+            ctx,
+            on_ended_closures,
+        })
+    }
+}
+
+impl StreamTrait for Stream {
+    fn play(&self) -> Result<(), PlayStreamError> {
+        let window = web_sys::window().unwrap();
+        match self.ctx.resume() {
+            Ok(_) => {
+                // Begin webaudio playback, initially scheduling the closures to fire on a timeout event.
+                let mut offset_ms = 10;
+                for on_ended_closure in self.on_ended_closures.iter() {
+                    window
+                        .set_timeout_with_callback_and_timeout_and_arguments_0(
+                            on_ended_closure
+                                .read()
+                                .unwrap()
+                                .as_ref()
+                                .unwrap()
+                                .as_ref()
+                                .unchecked_ref(),
+                            offset_ms,
+                        )
+                        .unwrap();
+                    offset_ms += 333 / 2;
+                }
+                Ok(())
+            }
+            Err(err) => {
+                let description = format!("{:?}", err);
+                let err = BackendSpecificError { description };
+                Err(err.into())
+            }
+        }
+    }
+
+    fn pause(&self) -> Result<(), PauseStreamError> {
+        match self.ctx.suspend() {
+            Ok(_) => Ok(()),
+            Err(err) => {
+                let description = format!("{:?}", err);
+                let err = BackendSpecificError { description };
+                Err(err.into())
+            }
+        }
+    }
+}
+
+impl Drop for Stream {
+    fn drop(&mut self) {
+        let _ = self.ctx.close();
+    }
+}
+
+impl Default for Devices {
+    fn default() -> Devices {
+        // We produce an empty iterator if the WebAudio API isn't available.
+        Devices(is_webaudio_available())
+    }
+}
+
+impl Iterator for Devices {
+    type Item = Device;
+    #[inline]
+    fn next(&mut self) -> Option<Device> {
+        if self.0 {
+            self.0 = false;
+            Some(Device)
+        } else {
+            None
+        }
+    }
+}
+
+#[inline]
+fn default_input_device() -> Option<Device> {
+    unimplemented!();
+}
+
+#[inline]
+fn default_output_device() -> Option<Device> {
+    if is_webaudio_available() {
+        Some(Device)
+    } else {
+        None
+    }
+}
+
+// Detects whether the `AudioContext` global variable is available.
+fn is_webaudio_available() -> bool {
+    if let Ok(audio_context_is_defined) = eval("typeof AudioContext !== 'undefined'") {
+        audio_context_is_defined.as_bool().unwrap()
+    } else {
+        false
+    }
+}

src/lib.rs

@@ -341,12 +341,14 @@ impl StreamInstant {
         (self.secs as i128 * 1_000_000_000) + self.nanos as i128
     }
 
+    #[allow(dead_code)]
     fn from_nanos(nanos: i64) -> Self {
         let secs = nanos / 1_000_000_000;
         let subsec_nanos = nanos - secs * 1_000_000_000;
         Self::new(secs as i64, subsec_nanos as u32)
     }
 
+    #[allow(dead_code)]
     fn from_nanos_i128(nanos: i128) -> Option<Self> {
         let secs = nanos / 1_000_000_000;
         if secs > std::i64::MAX as i128 || secs < std::i64::MIN as i128 {
@@ -358,6 +360,13 @@ impl StreamInstant {
         }
     }
 
+    #[allow(dead_code)]
+    fn from_secs_f64(secs: f64) -> crate::StreamInstant {
+        let s = secs.floor() as i64;
+        let ns = ((secs - s as f64) * 1_000_000_000.0) as u32;
+        Self::new(s, ns)
+    }
+
     fn new(secs: i64, nanos: u32) -> Self {
         StreamInstant { secs, nanos }
     }

src/platform/mod.rs

@@ -500,6 +500,24 @@ mod platform_impl {
     }
 }
 
+#[cfg(all(target_arch = "wasm32", feature = "wasm-bindgen"))]
+mod platform_impl {
+    pub use crate::host::webaudio::{
+        Device as WebAudioDevice, Devices as WebAudioDevices, Host as WebAudioHost,
+        Stream as WebAudioStream, SupportedInputConfigs as WebAudioSupportedInputConfigs,
+        SupportedOutputConfigs as WebAudioSupportedOutputConfigs,
+    };
+
+    impl_platform_host!(WebAudio webaudio "WebAudio");
+
+    /// The default host for the current compilation target platform.
+    pub fn default_host() -> Host {
+        WebAudioHost::new()
+            .expect("the default host should always be available")
+            .into()
+    }
+}
+
 #[cfg(windows)]
 mod platform_impl {
     #[cfg(feature = "asio")]
@@ -535,7 +553,8 @@ mod platform_impl {
     target_os = "freebsd",
     target_os = "macos",
     target_os = "ios",
-    target_os = "emscripten"
+    target_os = "emscripten",
+    all(target_arch = "wasm32", feature = "wasm-bindgen"),
 )))]
 mod platform_impl {
     pub use crate::host::null::{