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//! The `AudioContext` type and constructor options
use std::error::Error;
use std::sync::Mutex;
use crate::context::{AudioContextState, BaseAudioContext, ConcreteBaseAudioContext};
use crate::events::{EventDispatch, EventHandler, EventLoop, EventPayload, EventType};
use crate::io::{self, AudioBackendManager, ControlThreadInit, NoneBackend, RenderThreadInit};
use crate::media_devices::{enumerate_devices_sync, MediaDeviceInfoKind};
use crate::media_streams::{MediaStream, MediaStreamTrack};
use crate::message::{ControlMessage, OneshotNotify};
use crate::node::{self, AudioNodeOptions};
use crate::render::graph::Graph;
use crate::MediaElement;
use crate::{AudioRenderCapacity, Event};
use futures_channel::oneshot;
/// Check if the provided sink_id is available for playback
///
/// It should be "", "none" or a valid output `sinkId` returned from [`enumerate_devices_sync`]
fn is_valid_sink_id(sink_id: &str) -> bool {
if sink_id.is_empty() || sink_id == "none" {
true
} else {
enumerate_devices_sync()
.into_iter()
.filter(|d| d.kind() == MediaDeviceInfoKind::AudioOutput)
.any(|d| d.device_id() == sink_id)
}
}
/// Identify the type of playback, which affects tradeoffs
/// between audio output latency and power consumption
#[derive(Copy, Clone, Debug)]
pub enum AudioContextLatencyCategory {
/// Balance audio output latency and power consumption.
Balanced,
/// Provide the lowest audio output latency possible without glitching. This is the default.
Interactive,
/// Prioritize sustained playback without interruption over audio output latency.
///
/// Lowest power consumption.
Playback,
/// Specify the number of seconds of latency
///
/// This latency is not guaranteed to be applied, it depends on the audio hardware capabilities
Custom(f64),
}
impl Default for AudioContextLatencyCategory {
fn default() -> Self {
Self::Interactive
}
}
#[derive(Copy, Clone, Debug)]
#[non_exhaustive]
/// This allows users to ask for a particular render quantum size.
///
/// Currently, only the default value is available
pub enum AudioContextRenderSizeCategory {
/// The default value of 128 frames
Default,
}
impl Default for AudioContextRenderSizeCategory {
fn default() -> Self {
Self::Default
}
}
/// Specify the playback configuration for the [`AudioContext`] constructor.
///
/// All fields are optional and will default to the value best suited for interactive playback on
/// your hardware configuration.
///
/// For future compatibility, it is best to construct a default implementation of this struct and
/// set the fields you would like to override:
/// ```
/// use web_audio_api::context::AudioContextOptions;
///
/// // Request a sample rate of 44.1 kHz, leave other fields to their default values
/// let opts = AudioContextOptions {
/// sample_rate: Some(44100.),
/// ..AudioContextOptions::default()
/// };
#[derive(Clone, Debug, Default)]
pub struct AudioContextOptions {
/// Identify the type of playback, which affects tradeoffs between audio output latency and
/// power consumption.
pub latency_hint: AudioContextLatencyCategory,
/// Sample rate of the audio context and audio output hardware. Use `None` for a default value.
pub sample_rate: Option<f32>,
/// The audio output device
/// - use `""` for the default audio output device
/// - use `"none"` to process the audio graph without playing through an audio output device.
/// - use `"sinkId"` to use the specified audio sink id, obtained with [`enumerate_devices_sync`]
pub sink_id: String,
/// Option to request a default, optimized or specific render quantum size. It is a hint that might not be honored.
pub render_size_hint: AudioContextRenderSizeCategory,
}
/// This interface represents an audio graph whose `AudioDestinationNode` is routed to a real-time
/// output device that produces a signal directed at the user.
// the naming comes from the web audio specification
#[allow(clippy::module_name_repetitions)]
pub struct AudioContext {
/// represents the underlying `BaseAudioContext`
base: ConcreteBaseAudioContext,
/// audio backend (play/pause functionality)
backend_manager: Mutex<Box<dyn AudioBackendManager>>,
/// Provider for rendering performance metrics
render_capacity: AudioRenderCapacity,
/// Initializer for the render thread (when restart is required)
render_thread_init: RenderThreadInit,
}
impl std::fmt::Debug for AudioContext {
fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
f.debug_struct("AudioContext")
.field("sink_id", &self.sink_id())
.field("base_latency", &self.base_latency())
.field("output_latency", &self.output_latency())
.field("base", &self.base())
.finish_non_exhaustive()
}
}
impl Drop for AudioContext {
fn drop(&mut self) {
// Continue playing the stream if the AudioContext goes out of scope
if self.state() == AudioContextState::Running {
let tombstone = Box::new(NoneBackend::void());
let original = std::mem::replace(self.backend_manager.get_mut().unwrap(), tombstone);
Box::leak(original);
}
}
}
impl BaseAudioContext for AudioContext {
fn base(&self) -> &ConcreteBaseAudioContext {
&self.base
}
}
impl Default for AudioContext {
fn default() -> Self {
Self::new(AudioContextOptions::default())
}
}
impl AudioContext {
/// Creates and returns a new `AudioContext` object.
///
/// This will play live audio on the default output device.
///
/// ```no_run
/// use web_audio_api::context::{AudioContext, AudioContextOptions};
///
/// // Request a sample rate of 44.1 kHz and default latency (buffer size 128, if available)
/// let opts = AudioContextOptions {
/// sample_rate: Some(44100.),
/// ..AudioContextOptions::default()
/// };
///
/// // Setup the audio context that will emit to your speakers
/// let context = AudioContext::new(opts);
///
/// // Alternatively, use the default constructor to get the best settings for your hardware
/// // let context = AudioContext::default();
/// ```
///
/// # Panics
///
/// The `AudioContext` constructor will panic when an invalid `sinkId` is provided in the
/// `AudioContextOptions`. In a future version, a `try_new` constructor will be introduced that
/// never panics.
#[allow(clippy::needless_pass_by_value)]
#[must_use]
pub fn new(mut options: AudioContextOptions) -> Self {
// Log, but ignore invalid sinks
if !is_valid_sink_id(&options.sink_id) {
log::error!("NotFoundError: invalid sinkId {:?}", options.sink_id);
options.sink_id = String::from("");
}
// Set up the audio output thread
let (control_thread_init, render_thread_init) = io::thread_init();
let backend = io::build_output(options, render_thread_init.clone());
let ControlThreadInit {
state,
frames_played,
ctrl_msg_send,
load_value_recv,
event_send,
event_recv,
} = control_thread_init;
// Construct the audio Graph and hand it to the render thread
let (node_id_producer, node_id_consumer) = llq::Queue::new().split();
let graph = Graph::new(node_id_producer);
let message = ControlMessage::Startup { graph };
ctrl_msg_send.send(message).unwrap();
// Set up the event loop thread that handles the events spawned by the render thread
let event_loop = EventLoop::new(event_recv);
// Put everything together in the BaseAudioContext (shared with offline context)
let base = ConcreteBaseAudioContext::new(
backend.sample_rate(),
backend.number_of_channels(),
state,
frames_played,
ctrl_msg_send,
event_send,
event_loop.clone(),
false,
node_id_consumer,
);
// Setup AudioRenderCapacity for this context
let base_clone = base.clone();
let render_capacity = AudioRenderCapacity::new(base_clone, load_value_recv);
// As the final step, spawn a thread for the event loop. If we do this earlier we may miss
// event handling of the initial events that are emitted right after render thread
// construction.
event_loop.run_in_thread();
Self {
base,
backend_manager: Mutex::new(backend),
render_capacity,
render_thread_init,
}
}
/// This represents the number of seconds of processing latency incurred by
/// the `AudioContext` passing the audio from the `AudioDestinationNode`
/// to the audio subsystem.
// We don't do any buffering between rendering the audio and sending
// it to the audio subsystem, so this value is zero. (see Gecko)
#[allow(clippy::unused_self)]
#[must_use]
pub fn base_latency(&self) -> f64 {
0.
}
/// The estimation in seconds of audio output latency, i.e., the interval
/// between the time the UA requests the host system to play a buffer and
/// the time at which the first sample in the buffer is actually processed
/// by the audio output device.
#[must_use]
#[allow(clippy::missing_panics_doc)]
pub fn output_latency(&self) -> f64 {
self.backend_manager.lock().unwrap().output_latency()
}
/// Identifier or the information of the current audio output device.
///
/// The initial value is `""`, which means the default audio output device.
#[allow(clippy::missing_panics_doc)]
pub fn sink_id(&self) -> String {
self.backend_manager.lock().unwrap().sink_id().to_owned()
}
/// Returns an [`AudioRenderCapacity`] instance associated with an AudioContext.
#[must_use]
pub fn render_capacity(&self) -> AudioRenderCapacity {
self.render_capacity.clone()
}
/// Update the current audio output device.
///
/// The provided `sink_id` string must match a device name `enumerate_devices_sync`.
///
/// Supplying `"none"` for the `sink_id` will process the audio graph without playing through an
/// audio output device.
///
/// This function operates synchronously and might block the current thread. An async version
/// is currently not implemented.
#[allow(clippy::needless_collect, clippy::missing_panics_doc)]
pub fn set_sink_id_sync(&self, sink_id: String) -> Result<(), Box<dyn Error>> {
log::debug!("SinkChange requested");
if self.sink_id() == sink_id {
log::debug!("SinkChange: no-op");
return Ok(()); // sink is already active
}
if !is_valid_sink_id(&sink_id) {
Err(format!("NotFoundError: invalid sinkId {sink_id}"))?;
};
log::debug!("SinkChange: locking backend manager");
let mut backend_manager_guard = self.backend_manager.lock().unwrap();
let original_state = self.state();
if original_state == AudioContextState::Closed {
log::debug!("SinkChange: context is closed");
return Ok(());
}
// Acquire exclusive lock on ctrl msg sender
log::debug!("SinkChange: locking message channel");
let ctrl_msg_send = self.base.lock_control_msg_sender();
// Flush out the ctrl msg receiver, cache
let mut pending_msgs: Vec<_> = self.render_thread_init.ctrl_msg_recv.try_iter().collect();
// Acquire the active audio graph from the current render thread, shutting it down
let graph = if matches!(pending_msgs.first(), Some(ControlMessage::Startup { .. })) {
// Handle the edge case where the previous backend was suspended for its entire lifetime.
// In this case, the `Startup` control message was never processed.
log::debug!("SinkChange: recover unstarted graph");
let msg = pending_msgs.remove(0);
match msg {
ControlMessage::Startup { graph } => graph,
_ => unreachable!(),
}
} else {
// Acquire the audio graph from the current render thread, shutting it down
log::debug!("SinkChange: recover graph from render thread");
let (graph_send, graph_recv) = crossbeam_channel::bounded(1);
let message = ControlMessage::CloseAndRecycle { sender: graph_send };
ctrl_msg_send.send(message).unwrap();
if original_state == AudioContextState::Suspended {
// We must wake up the render thread to be able to handle the shutdown.
// No new audio will be produced because it will receive the shutdown command first.
backend_manager_guard.resume();
}
graph_recv.recv().unwrap()
};
log::debug!("SinkChange: closing audio stream");
backend_manager_guard.close();
// hotswap the backend
let options = AudioContextOptions {
sample_rate: Some(self.sample_rate()),
latency_hint: AudioContextLatencyCategory::default(), // todo reuse existing setting
sink_id,
render_size_hint: AudioContextRenderSizeCategory::default(), // todo reuse existing setting
};
log::debug!("SinkChange: starting audio stream");
*backend_manager_guard = io::build_output(options, self.render_thread_init.clone());
// if the previous backend state was suspend, suspend the new one before shipping the graph
if original_state == AudioContextState::Suspended {
log::debug!("SinkChange: suspending audio stream");
backend_manager_guard.suspend();
}
// send the audio graph to the new render thread
let message = ControlMessage::Startup { graph };
ctrl_msg_send.send(message).unwrap();
// flush the cached msgs
pending_msgs
.into_iter()
.for_each(|m| self.base().send_control_msg(m));
// explicitly release the lock to prevent concurrent render threads
drop(backend_manager_guard);
// trigger event when all the work is done
let _ = self.base.send_event(EventDispatch::sink_change());
log::debug!("SinkChange: done");
Ok(())
}
/// Register callback to run when the audio sink has changed
///
/// Only a single event handler is active at any time. Calling this method multiple times will
/// override the previous event handler.
pub fn set_onsinkchange<F: FnMut(Event) + Send + 'static>(&self, mut callback: F) {
let callback = move |_| {
callback(Event {
type_: "sinkchange",
})
};
self.base().set_event_handler(
EventType::SinkChange,
EventHandler::Multiple(Box::new(callback)),
);
}
/// Unset the callback to run when the audio sink has changed
pub fn clear_onsinkchange(&self) {
self.base().clear_event_handler(EventType::SinkChange);
}
#[allow(clippy::missing_panics_doc)]
#[doc(hidden)] // Method signature might change in the future
pub fn run_diagnostics<F: Fn(String) + Send + 'static>(&self, callback: F) {
let mut buffer = Vec::with_capacity(32 * 1024);
{
let backend = self.backend_manager.lock().unwrap();
use std::io::Write;
writeln!(&mut buffer, "backend: {}", backend.name()).ok();
writeln!(&mut buffer, "sink id: {}", backend.sink_id()).ok();
writeln!(
&mut buffer,
"output latency: {:.6}",
backend.output_latency()
)
.ok();
}
let callback = move |v| match v {
EventPayload::Diagnostics(v) => {
let s = String::from_utf8(v).unwrap();
callback(s);
}
_ => unreachable!(),
};
self.base().set_event_handler(
EventType::Diagnostics,
EventHandler::Once(Box::new(callback)),
);
self.base()
.send_control_msg(ControlMessage::RunDiagnostics { buffer });
}
/// Suspends the progression of time in the audio context.
///
/// This will temporarily halt audio hardware access and reducing CPU/battery usage in the
/// process.
///
/// # Panics
///
/// Will panic if:
///
/// * The audio device is not available
/// * For a `BackendSpecificError`
pub async fn suspend(&self) {
// Don't lock the backend manager because we can't hold is across the await point
log::debug!("Suspend called");
if self.state() != AudioContextState::Running {
log::debug!("Suspend no-op - context is not running");
return;
}
// Pause rendering via a control message
let (sender, receiver) = oneshot::channel();
let notify = OneshotNotify::Async(sender);
self.base
.send_control_msg(ControlMessage::Suspend { notify });
// Wait for the render thread to have processed the suspend message.
// The AudioContextState will be updated by the render thread.
log::debug!("Suspending audio graph, waiting for signal..");
receiver.await.unwrap();
// Then ask the audio host to suspend the stream
log::debug!("Suspended audio graph. Suspending audio stream..");
self.backend_manager.lock().unwrap().suspend();
log::debug!("Suspended audio stream");
}
/// Resumes the progression of time in an audio context that has previously been
/// suspended/paused.
///
/// # Panics
///
/// Will panic if:
///
/// * The audio device is not available
/// * For a `BackendSpecificError`
pub async fn resume(&self) {
let (sender, receiver) = oneshot::channel();
{
// Lock the backend manager mutex to avoid concurrent calls
log::debug!("Resume called, locking backend manager");
let backend_manager_guard = self.backend_manager.lock().unwrap();
if self.state() != AudioContextState::Suspended {
log::debug!("Resume no-op - context is not suspended");
return;
}
// Ask the audio host to resume the stream
backend_manager_guard.resume();
// Then, ask to resume rendering via a control message
log::debug!("Resumed audio stream, waking audio graph");
let notify = OneshotNotify::Async(sender);
self.base
.send_control_msg(ControlMessage::Resume { notify });
// Drop the Mutex guard so we won't hold it across an await point
}
// Wait for the render thread to have processed the resume message
// The AudioContextState will be updated by the render thread.
receiver.await.unwrap();
log::debug!("Resumed audio graph");
}
/// Closes the `AudioContext`, releasing the system resources being used.
///
/// This will not automatically release all `AudioContext`-created objects, but will suspend
/// the progression of the currentTime, and stop processing audio data.
///
/// # Panics
///
/// Will panic when this function is called multiple times
pub async fn close(&self) {
// Don't lock the backend manager because we can't hold is across the await point
log::debug!("Close called");
if self.state() == AudioContextState::Closed {
log::debug!("Close no-op - context is already closed");
return;
}
if self.state() == AudioContextState::Running {
// First, stop rendering via a control message
let (sender, receiver) = oneshot::channel();
let notify = OneshotNotify::Async(sender);
self.base.send_control_msg(ControlMessage::Close { notify });
// Wait for the render thread to have processed the suspend message.
// The AudioContextState will be updated by the render thread.
log::debug!("Suspending audio graph, waiting for signal..");
receiver.await.unwrap();
} else {
// if the context is not running, change the state manually
self.base.set_state(AudioContextState::Closed);
}
// Then ask the audio host to close the stream
log::debug!("Suspended audio graph. Closing audio stream..");
self.backend_manager.lock().unwrap().close();
// Stop the AudioRenderCapacity collection thread
self.render_capacity.stop();
log::debug!("Closed audio stream");
}
/// Suspends the progression of time in the audio context.
///
/// This will temporarily halt audio hardware access and reducing CPU/battery usage in the
/// process.
///
/// This function operates synchronously and blocks the current thread until the audio thread
/// has stopped processing.
///
/// # Panics
///
/// Will panic if:
///
/// * The audio device is not available
/// * For a `BackendSpecificError`
pub fn suspend_sync(&self) {
// Lock the backend manager mutex to avoid concurrent calls
log::debug!("Suspend_sync called, locking backend manager");
let backend_manager_guard = self.backend_manager.lock().unwrap();
if self.state() != AudioContextState::Running {
log::debug!("Suspend_sync no-op - context is not running");
return;
}
// Pause rendering via a control message
let (sender, receiver) = crossbeam_channel::bounded(0);
let notify = OneshotNotify::Sync(sender);
self.base
.send_control_msg(ControlMessage::Suspend { notify });
// Wait for the render thread to have processed the suspend message.
// The AudioContextState will be updated by the render thread.
log::debug!("Suspending audio graph, waiting for signal..");
receiver.recv().ok();
// Then ask the audio host to suspend the stream
log::debug!("Suspended audio graph. Suspending audio stream..");
backend_manager_guard.suspend();
log::debug!("Suspended audio stream");
}
/// Resumes the progression of time in an audio context that has previously been
/// suspended/paused.
///
/// This function operates synchronously and blocks the current thread until the audio thread
/// has started processing again.
///
/// # Panics
///
/// Will panic if:
///
/// * The audio device is not available
/// * For a `BackendSpecificError`
pub fn resume_sync(&self) {
// Lock the backend manager mutex to avoid concurrent calls
log::debug!("Resume_sync called, locking backend manager");
let backend_manager_guard = self.backend_manager.lock().unwrap();
if self.state() != AudioContextState::Suspended {
log::debug!("Resume no-op - context is not suspended");
return;
}
// Ask the audio host to resume the stream
backend_manager_guard.resume();
// Then, ask to resume rendering via a control message
log::debug!("Resumed audio stream, waking audio graph");
let (sender, receiver) = crossbeam_channel::bounded(0);
let notify = OneshotNotify::Sync(sender);
self.base
.send_control_msg(ControlMessage::Resume { notify });
// Wait for the render thread to have processed the resume message
// The AudioContextState will be updated by the render thread.
receiver.recv().ok();
log::debug!("Resumed audio graph");
}
/// Closes the `AudioContext`, releasing the system resources being used.
///
/// This will not automatically release all `AudioContext`-created objects, but will suspend
/// the progression of the currentTime, and stop processing audio data.
///
/// This function operates synchronously and blocks the current thread until the audio thread
/// has stopped processing.
///
/// # Panics
///
/// Will panic when this function is called multiple times
pub fn close_sync(&self) {
// Lock the backend manager mutex to avoid concurrent calls
log::debug!("Close_sync called, locking backend manager");
let backend_manager_guard = self.backend_manager.lock().unwrap();
if self.state() == AudioContextState::Closed {
log::debug!("Close no-op - context is already closed");
return;
}
// First, stop rendering via a control message
if self.state() == AudioContextState::Running {
let (sender, receiver) = crossbeam_channel::bounded(0);
let notify = OneshotNotify::Sync(sender);
self.base.send_control_msg(ControlMessage::Close { notify });
// Wait for the render thread to have processed the suspend message.
// The AudioContextState will be updated by the render thread.
log::debug!("Suspending audio graph, waiting for signal..");
receiver.recv().ok();
} else {
// if the context is not running, change the state manually
self.base.set_state(AudioContextState::Closed);
}
// Then ask the audio host to close the stream
log::debug!("Suspended audio graph. Closing audio stream..");
backend_manager_guard.close();
// Stop the AudioRenderCapacity collection thread
self.render_capacity.stop();
log::debug!("Closed audio stream");
}
/// Creates a [`MediaStreamAudioSourceNode`](node::MediaStreamAudioSourceNode) from a
/// [`MediaStream`]
#[must_use]
pub fn create_media_stream_source(
&self,
media: &MediaStream,
) -> node::MediaStreamAudioSourceNode {
let opts = node::MediaStreamAudioSourceOptions {
media_stream: media,
};
node::MediaStreamAudioSourceNode::new(self, opts)
}
/// Creates a [`MediaStreamAudioDestinationNode`](node::MediaStreamAudioDestinationNode)
#[must_use]
pub fn create_media_stream_destination(&self) -> node::MediaStreamAudioDestinationNode {
let opts = AudioNodeOptions::default();
node::MediaStreamAudioDestinationNode::new(self, opts)
}
/// Creates a [`MediaStreamTrackAudioSourceNode`](node::MediaStreamTrackAudioSourceNode) from a
/// [`MediaStreamTrack`]
#[must_use]
pub fn create_media_stream_track_source(
&self,
media: &MediaStreamTrack,
) -> node::MediaStreamTrackAudioSourceNode {
let opts = node::MediaStreamTrackAudioSourceOptions {
media_stream_track: media,
};
node::MediaStreamTrackAudioSourceNode::new(self, opts)
}
/// Creates a [`MediaElementAudioSourceNode`](node::MediaElementAudioSourceNode) from a
/// [`MediaElement`]
#[must_use]
pub fn create_media_element_source(
&self,
media_element: &mut MediaElement,
) -> node::MediaElementAudioSourceNode {
let opts = node::MediaElementAudioSourceOptions { media_element };
node::MediaElementAudioSourceNode::new(self, opts)
}
}
#[cfg(test)]
mod tests {
use super::*;
use futures::executor;
#[test]
fn test_suspend_resume_close() {
let options = AudioContextOptions {
sink_id: "none".into(),
..AudioContextOptions::default()
};
// construct with 'none' sink_id
let context = AudioContext::new(options);
// allow some time to progress
std::thread::sleep(std::time::Duration::from_millis(1));
executor::block_on(context.suspend());
assert_eq!(context.state(), AudioContextState::Suspended);
let time1 = context.current_time();
assert!(time1 >= 0.);
// allow some time to progress
std::thread::sleep(std::time::Duration::from_millis(1));
let time2 = context.current_time();
assert_eq!(time1, time2); // no progression of time
executor::block_on(context.resume());
assert_eq!(context.state(), AudioContextState::Running);
// allow some time to progress
std::thread::sleep(std::time::Duration::from_millis(1));
let time3 = context.current_time();
assert!(time3 > time2); // time is progressing
executor::block_on(context.close());
assert_eq!(context.state(), AudioContextState::Closed);
let time4 = context.current_time();
// allow some time to progress
std::thread::sleep(std::time::Duration::from_millis(1));
let time5 = context.current_time();
assert_eq!(time5, time4); // no progression of time
}
fn require_send_sync<T: Send + Sync>(_: T) {}
#[test]
fn test_all_futures_thread_safe() {
let options = AudioContextOptions {
sink_id: "none".into(),
..AudioContextOptions::default()
};
let context = AudioContext::new(options);
require_send_sync(context.suspend());
require_send_sync(context.resume());
require_send_sync(context.close());
}
}