webaudio: Fix default_output_config()

Fix compilation error when using null host

.github/workflows/cpal.yml

@@ -127,6 +127,25 @@ jobs:
       - name: Build beep example
         run: cargo build --example beep --target ${{ matrix.target }} --features=wasm-bindgen
 
+  wasm32-wasi-test:
+
+    strategy:
+      matrix:
+        target: [wasm32-wasi]
+
+    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 }}
+
   windows-test:
     strategy:
       matrix:

CHANGELOG.md

@@ -1,4 +1,8 @@
-# Unreleased
+# Version 0.12.1 (2020-07-23)
+
+- Bugfix release to get the asio feature working again.
+
+# Version 0.12.0 (2020-07-09)
 
 - Large refactor removing the blocking EventLoop API.
 - Rename many `Format` types to `StreamConfig`:

Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "cpal"
-version = "0.11.0"
+version = "0.12.1"
 authors = ["The CPAL contributors", "Pierre Krieger <pierre.krieger1708@gmail.com>"]
 description = "Low-level cross-platform audio I/O library in pure Rust."
 repository = "https://github.com/rustaudio/cpal"
@@ -9,12 +9,10 @@ license = "Apache-2.0"
 keywords = ["audio", "sound"]
 
 [features]
-asio = ["asio-sys"] # Only available on Windows. See README for setup instructions.
+asio = ["asio-sys", "num-traits"] # Only available on Windows. See README for setup instructions.
 
 [dependencies]
 thiserror = "1.0.2"
-lazy_static = "1.3"
-num-traits = "0.2.6"
 
 [dev-dependencies]
 anyhow = "1.0.12"
@@ -23,8 +21,10 @@ ringbuf = "0.1.6"
 
 [target.'cfg(target_os = "windows")'.dependencies]
 winapi = { version = "0.3", features = ["audiosessiontypes", "audioclient", "coml2api", "combaseapi", "debug", "devpkey", "handleapi", "ksmedia", "mmdeviceapi", "objbase", "profileapi", "std", "synchapi", "winbase", "winuser"] }
-asio-sys = { version = "0.1", path = "asio-sys", optional = true }
+asio-sys = { version = "0.2", path = "asio-sys", optional = true }
+num-traits = { version = "0.2.6", optional = true }
 parking_lot = "0.9"
+lazy_static = "1.3"
 
 [target.'cfg(any(target_os = "linux", target_os = "dragonfly", target_os = "freebsd"))'.dependencies]
 alsa = "0.4.1"

README.md

@@ -26,6 +26,10 @@ Note that on Linux, the ALSA development files are required. These are provided
 as part of the `libasound2-dev` package on Debian and Ubuntu distributions and
 `alsa-lib-devel` on Fedora.
 
+## Compiling for Web Assembly
+
+If you are interested in using CPAL with WASM, please see [this guide](https://github.com/RustAudio/cpal/wiki/Setting-up-a-new-CPAL-WASM-project) in our Wiki which walks through setting up a new project from scratch. 
+
 ## ASIO on Windows
 
 [ASIO](https://en.wikipedia.org/wiki/Audio_Stream_Input/Output) is an audio

asio-sys/Cargo.toml

@@ -1,20 +1,20 @@
 [package]
 name = "asio-sys"
-version = "0.1.0"
+version = "0.2.0"
 authors = ["Tom Gowan <tomrgowan@gmail.com>"]
 description = "Low-level interface and binding generation for the steinberg ASIO SDK."
-repository = "https://github.com/tomaka/cpal"
+repository = "https://github.com/RustAudio/cpal/"
 documentation = "https://docs.rs/asio-sys"
 license = "Apache-2.0"
 keywords = ["audio", "sound", "asio", "steinberg"]
 build = "build.rs"
 
 [target.'cfg(any(target_os = "windows"))'.build-dependencies]
-bindgen = "0.51.0"
+bindgen = "0.54.0"
 walkdir = "2"
 cc = "1.0.25"
 
 [dependencies]
 lazy_static = "1.0.0"
-num-derive = "0.2"
+num-derive = "0.3"
 num-traits = "0.2"

asio-sys/src/bindings/errors.rs

@@ -19,6 +19,7 @@ pub enum AsioError {
     HardwareStuck,
     NoRate,
     ASE_NoMemory,
+    InvalidBufferSize,
     UnknownError,
 }
 
@@ -63,6 +64,7 @@ impl fmt::Display for AsioError {
                 "sample clock or rate cannot be determined or is not present"
             ),
             AsioError::ASE_NoMemory => write!(f, "not enough memory for completing the request"),
+            AsioError::InvalidBufferSize => write!(f, "buffersize out of range for device"),
             AsioError::UnknownError => write!(f, "Error not in SDK"),
         }
     }
@@ -94,6 +96,7 @@ impl Error for AsioError {
             AsioError::HardwareStuck => "hardware is not running when sample position is inquired",
             AsioError::NoRate => "sample clock or rate cannot be determined or is not present",
             AsioError::ASE_NoMemory => "not enough memory for completing the request",
+            AsioError::InvalidBufferSize => "buffersize out of range for device",
             AsioError::UnknownError => "Error not in SDK",
         }
     }

asio-sys/src/bindings/mod.rs

@@ -385,6 +385,14 @@ impl Driver {
         Ok(channel)
     }
 
+    /// Get the min and max supported buffersize of the driver.
+    pub fn buffersize_range(&self) -> Result<(c_long, c_long), AsioError> {
+        let buffer_sizes = asio_get_buffer_sizes()?;
+        let min = buffer_sizes.min;
+        let max = buffer_sizes.max;
+        Ok((min, max))
+    }
+
     /// Get current sample rate of the driver.
     pub fn sample_rate(&self) -> Result<c_double, AsioError> {
         let mut rate: c_double = 0.0;
@@ -431,8 +439,14 @@ impl Driver {
     ///
     /// This will destroy any already allocated buffers.
     ///
-    /// The preferred buffer size from ASIO is used.
+    /// If buffersize is None then the preferred buffer size from ASIO is used,
-    fn create_buffers(&self, buffer_infos: &mut [AsioBufferInfo]) -> Result<c_long, AsioError> {
+    /// otherwise the desired buffersize is used if the requeted size is within
+    /// the range of accepted buffersizes for the device.
+    fn create_buffers(
+        &self,
+        buffer_infos: &mut [AsioBufferInfo],
+        buffer_size: Option<i32>,
+    ) -> Result<c_long, AsioError> {
         let num_channels = buffer_infos.len();
 
         // To pass as ai::ASIOCallbacks
@@ -449,6 +463,17 @@ impl Driver {
             );
         }
 
+        let buffer_size = match buffer_size {
+            Some(v) => {
+                if v <= buffer_sizes.max {
+                    v
+                } else {
+                    return Err(AsioError::InvalidBufferSize);
+                }
+            }
+            None => buffer_sizes.pref,
+        };
+
         // Ensure the driver is in the `Initialized` state.
         if let DriverState::Running = *state {
             state.stop()?;
@@ -460,23 +485,27 @@ impl Driver {
             asio_result!(ai::ASIOCreateBuffers(
                 buffer_infos.as_mut_ptr() as *mut _,
                 num_channels as i32,
-                buffer_sizes.pref,
+                buffer_size,
                 &mut callbacks as *mut _ as *mut _,
             ))?;
         }
         *state = DriverState::Prepared;
 
-        Ok(buffer_sizes.pref)
+        Ok(buffer_size)
     }
 
     /// Creates the streams.
     ///
+    /// `buffer_size` sets the desired buffer_size. If None is passed in, then the
+    /// default buffersize for the device is used.
+    ///
     /// Both input and output streams need to be created together as a single slice of
     /// `ASIOBufferInfo`.
     fn create_streams(
         &self,
         mut input_buffer_infos: Vec<AsioBufferInfo>,
         mut output_buffer_infos: Vec<AsioBufferInfo>,
+        buffer_size: Option<i32>,
     ) -> Result<AsioStreams, AsioError> {
         let (input, output) = match (
             input_buffer_infos.is_empty(),
@@ -489,7 +518,7 @@ impl Driver {
                 let mut all_buffer_infos = input_buffer_infos;
                 all_buffer_infos.append(&mut output_buffer_infos);
                 // Create the buffers. On success, split the output and input again.
-                let buffer_size = self.create_buffers(&mut all_buffer_infos)?;
+                let buffer_size = self.create_buffers(&mut all_buffer_infos, buffer_size)?;
                 let output_buffer_infos = all_buffer_infos.split_off(split_point);
                 let input_buffer_infos = all_buffer_infos;
                 let input = Some(AsioStream {
@@ -504,7 +533,7 @@ impl Driver {
             }
             // Just input
             (false, true) => {
-                let buffer_size = self.create_buffers(&mut input_buffer_infos)?;
+                let buffer_size = self.create_buffers(&mut input_buffer_infos, buffer_size)?;
                 let input = Some(AsioStream {
                     buffer_infos: input_buffer_infos,
                     buffer_size,
@@ -514,7 +543,7 @@ impl Driver {
             }
             // Just output
             (true, false) => {
-                let buffer_size = self.create_buffers(&mut output_buffer_infos)?;
+                let buffer_size = self.create_buffers(&mut output_buffer_infos, buffer_size)?;
                 let input = None;
                 let output = Some(AsioStream {
                     buffer_infos: output_buffer_infos,
@@ -537,17 +566,21 @@ impl Driver {
     ///
     /// `num_channels` is the desired number of input channels.
     ///
+    /// `buffer_size` sets the desired buffer_size. If None is passed in, then the
+    /// default buffersize for the device is used.
+    ///
     /// This returns a full AsioStreams with both input and output if output was active.
     pub fn prepare_input_stream(
         &self,
         output: Option<AsioStream>,
         num_channels: usize,
+        buffer_size: Option<i32>,
     ) -> Result<AsioStreams, AsioError> {
         let input_buffer_infos = prepare_buffer_infos(true, num_channels);
         let output_buffer_infos = output
             .map(|output| output.buffer_infos)
             .unwrap_or_else(Vec::new);
-        self.create_streams(input_buffer_infos, output_buffer_infos)
+        self.create_streams(input_buffer_infos, output_buffer_infos, buffer_size)
     }
 
     /// Prepare the output stream.
@@ -559,17 +592,21 @@ impl Driver {
     ///
     /// `num_channels` is the desired number of output channels.
     ///
+    /// `buffer_size` sets the desired buffer_size. If None is passed in, then the
+    /// default buffersize for the device is used.
+    ///
     /// This returns a full AsioStreams with both input and output if input was active.
     pub fn prepare_output_stream(
         &self,
         input: Option<AsioStream>,
         num_channels: usize,
+        buffer_size: Option<i32>,
     ) -> Result<AsioStreams, AsioError> {
         let input_buffer_infos = input
             .map(|input| input.buffer_infos)
             .unwrap_or_else(Vec::new);
         let output_buffer_infos = prepare_buffer_infos(false, num_channels);
-        self.create_streams(input_buffer_infos, output_buffer_infos)
+        self.create_streams(input_buffer_infos, output_buffer_infos, buffer_size)
     }
 
     /// Releases buffers allocations.

examples/enumerate.rs

@@ -11,6 +11,7 @@ fn main() -> Result<(), anyhow::Error> {
     for host_id in available_hosts {
         println!("{}", host_id.name());
         let host = cpal::host_from_id(host_id)?;
+
         let default_in = host.default_input_device().map(|e| e.name().unwrap());
         let default_out = host.default_output_device().map(|e| e.name().unwrap());
         println!("  Default Input Device:\n    {:?}", default_in);

examples/wasm-beep/package-lock.json

@@ -3417,9 +3417,9 @@
       }
     },
     "lodash": {
-      "version": "4.17.15",
+      "version": "4.17.19",
-      "resolved": "https://registry.npmjs.org/lodash/-/lodash-4.17.15.tgz",
+      "resolved": "https://registry.npmjs.org/lodash/-/lodash-4.17.19.tgz",
-      "integrity": "sha512-8xOcRHvCjnocdS5cpwXQXVzmmh5e5+saE2QGoeQmbKmRS6J3VQppPOIt0MnmE+4xlZoumy0GPG0D0MVIQbNA1A=="
+      "integrity": "sha512-JNvd8XER9GQX0v2qJgsaN/mzFCNA5BRe/j8JN9d+tWyGLSodKQHKFicdwNYzWwI3wjRnaKPsGj1XkBjx/F96DQ=="
     },
     "loglevel": {
       "version": "1.6.7",

src/host/alsa/mod.rs

@@ -3,16 +3,17 @@ extern crate libc;
 
 use self::alsa::poll::Descriptors;
 use crate::{
-    BackendSpecificError, BuildStreamError, ChannelCount, Data, DefaultStreamConfigError,
+    BackendSpecificError, BufferSize, BuildStreamError, ChannelCount, Data,
-    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
+    DefaultStreamConfigError, DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo,
-    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
+    PauseStreamError, PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError,
-    SupportedStreamConfigRange, SupportedStreamConfigsError,
+    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
+    SupportedStreamConfigsError,
 };
+use std::cmp;
 use std::convert::TryInto;
 use std::sync::Arc;
 use std::thread::{self, JoinHandle};
 use std::vec::IntoIter as VecIntoIter;
-use std::{cmp, mem};
 use traits::{DeviceTrait, HostTrait, StreamTrait};
 
 pub use self::enumerate::{default_input_device, default_output_device, Devices};
@@ -339,6 +340,14 @@ impl Device {
             })
             .collect::<Vec<_>>();
 
+        let min_buffer_size = hw_params.get_buffer_size_min()?;
+        let max_buffer_size = hw_params.get_buffer_size_max()?;
+
+        let buffer_size_range = SupportedBufferSize::Range {
+            min: min_buffer_size as u32,
+            max: max_buffer_size as u32,
+        };
+
         let mut output = Vec::with_capacity(
             supported_formats.len() * supported_channels.len() * sample_rates.len(),
         );
@@ -349,6 +358,7 @@ impl Device {
                         channels: channels.clone(),
                         min_sample_rate: SampleRate(min_rate as u32),
                         max_sample_rate: SampleRate(max_rate as u32),
+                        buffer_size: buffer_size_range.clone(),
                         sample_format: sample_format,
                     });
                 }
@@ -438,6 +448,7 @@ struct StreamInner {
 
     #[allow(dead_code)]
     // Whether or not the hardware supports pausing the stream.
+    // TODO: We need an API to expose this. See #197, #284.
     can_pause: bool,
 
     // In the case that the device does not return valid timestamps via `get_htstamp`, this field
@@ -870,7 +881,7 @@ fn set_hw_params_from_format<'a>(
     config: &StreamConfig,
     sample_format: SampleFormat,
 ) -> Result<alsa::pcm::HwParams<'a>, BackendSpecificError> {
-    let mut hw_params = alsa::pcm::HwParams::any(pcm_handle)?;
+    let hw_params = alsa::pcm::HwParams::any(pcm_handle)?;
     hw_params.set_access(alsa::pcm::Access::RWInterleaved)?;
 
     let sample_format = if cfg!(target_endian = "big") {
@@ -891,11 +902,14 @@ fn set_hw_params_from_format<'a>(
     hw_params.set_rate(config.sample_rate.0, alsa::ValueOr::Nearest)?;
     hw_params.set_channels(config.channels as u32)?;
 
-    // If this isn't set manually a overlarge buffer may be used causing audio delay
+    match config.buffer_size {
-    let mut hw_params_copy = hw_params.clone();
+        BufferSize::Fixed(v) => hw_params.set_buffer_size(v as alsa::pcm::Frames)?,
-    if let Err(_) = hw_params.set_buffer_time_near(100_000, alsa::ValueOr::Nearest) {
+        BufferSize::Default => {
-        // Swap out the params with errors for a snapshot taken before the error was introduced.
+            // These values together represent a moderate latency and wakeup interval.
-        mem::swap(&mut hw_params_copy, &mut hw_params);
+            // Without them we are at the mercy of the device
+            hw_params.set_period_time_near(25_000, alsa::ValueOr::Nearest)?;
+            hw_params.set_buffer_time_near(100_000, alsa::ValueOr::Nearest)?;
+        }
     }
 
     pcm_handle.hw_params(&hw_params)?;

src/host/asio/device.rs

@@ -12,6 +12,7 @@ use DeviceNameError;
 use DevicesError;
 use SampleFormat;
 use SampleRate;
+use SupportedBufferSize;
 use SupportedStreamConfig;
 use SupportedStreamConfigRange;
 use SupportedStreamConfigsError;
@@ -77,9 +78,13 @@ impl Device {
                 continue;
             }
             for channels in 1..f.channels + 1 {
-                f.channels = channels;
+                supported_configs.push(SupportedStreamConfigRange {
-                f.sample_rate = rate;
+                    channels,
-                supported_configs.push(SupportedStreamConfigRange::from(f.clone()));
+                    min_sample_rate: rate,
+                    max_sample_rate: rate,
+                    buffer_size: f.buffer_size.clone(),
+                    sample_format: f.sample_format.clone(),
+                })
             }
         }
         Ok(supported_configs.into_iter())
@@ -110,9 +115,13 @@ impl Device {
                 continue;
             }
             for channels in 1..f.channels + 1 {
-                f.channels = channels;
+                supported_configs.push(SupportedStreamConfigRange {
-                f.sample_rate = rate;
+                    channels,
-                supported_configs.push(SupportedStreamConfigRange::from(f.clone()));
+                    min_sample_rate: rate,
+                    max_sample_rate: rate,
+                    buffer_size: f.buffer_size.clone(),
+                    sample_format: f.sample_format.clone(),
+                })
             }
         }
         Ok(supported_configs.into_iter())
@@ -122,6 +131,11 @@ impl Device {
     pub fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
         let channels = self.driver.channels().map_err(default_config_err)?.ins as u16;
         let sample_rate = SampleRate(self.driver.sample_rate().map_err(default_config_err)? as _);
+        let (min, max) = self.driver.buffersize_range().map_err(default_config_err)?;
+        let buffer_size = SupportedBufferSize::Range {
+            min: min as u32,
+            max: max as u32,
+        };
         // Map th ASIO sample type to a CPAL sample type
         let data_type = self.driver.input_data_type().map_err(default_config_err)?;
         let sample_format = convert_data_type(&data_type)
@@ -129,6 +143,7 @@ impl Device {
         Ok(SupportedStreamConfig {
             channels,
             sample_rate,
+            buffer_size,
             sample_format,
         })
     }
@@ -137,12 +152,18 @@ impl Device {
     pub fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
         let channels = self.driver.channels().map_err(default_config_err)?.outs as u16;
         let sample_rate = SampleRate(self.driver.sample_rate().map_err(default_config_err)? as _);
+        let (min, max) = self.driver.buffersize_range().map_err(default_config_err)?;
+        let buffer_size = SupportedBufferSize::Range {
+            min: min as u32,
+            max: max as u32,
+        };
         let data_type = self.driver.output_data_type().map_err(default_config_err)?;
         let sample_format = convert_data_type(&data_type)
             .ok_or(DefaultStreamConfigError::StreamTypeNotSupported)?;
         Ok(SupportedStreamConfig {
             channels,
             sample_rate,
+            buffer_size,
             sample_format,
         })
     }

src/host/asio/stream.rs

@@ -5,9 +5,9 @@ use self::num_traits::PrimInt;
 use super::parking_lot::Mutex;
 use super::Device;
 use crate::{
-    BackendSpecificError, BuildStreamError, Data, InputCallbackInfo, OutputCallbackInfo,
+    BackendSpecificError, BufferSize, BuildStreamError, Data, InputCallbackInfo,
-    PauseStreamError, PlayStreamError, Sample, SampleFormat, StreamConfig, StreamError,
+    OutputCallbackInfo, PauseStreamError, PlayStreamError, Sample, SampleFormat, StreamConfig,
-    SupportedStreamConfig,
+    StreamError,
 };
 use std;
 use std::sync::atomic::{AtomicBool, Ordering};
@@ -482,6 +482,12 @@ impl Device {
         }?;
         let num_channels = config.channels as usize;
         let ref mut streams = *self.asio_streams.lock();
+
+        let buffer_size = match config.buffer_size {
+            BufferSize::Fixed(v) => Some(v as i32),
+            BufferSize::Default => None,
+        };
+
         // Either create a stream if thers none or had back the
         // size of the current one.
         match streams.input {
@@ -489,7 +495,7 @@ impl Device {
             None => {
                 let output = streams.output.take();
                 self.driver
-                    .prepare_input_stream(output, num_channels)
+                    .prepare_input_stream(output, num_channels, buffer_size)
                     .map(|new_streams| {
                         let bs = match new_streams.input {
                             Some(ref inp) => inp.buffer_size as usize,
@@ -523,6 +529,12 @@ impl Device {
         }?;
         let num_channels = config.channels as usize;
         let ref mut streams = *self.asio_streams.lock();
+
+        let buffer_size = match config.buffer_size {
+            BufferSize::Fixed(v) => Some(v as i32),
+            BufferSize::Default => None,
+        };
+
         // Either create a stream if thers none or had back the
         // size of the current one.
         match streams.output {
@@ -530,7 +542,7 @@ impl Device {
             None => {
                 let output = streams.output.take();
                 self.driver
-                    .prepare_output_stream(output, num_channels)
+                    .prepare_output_stream(output, num_channels, buffer_size)
                     .map(|new_streams| {
                         let bs = match new_streams.output {
                             Some(ref out) => out.buffer_size as usize,
@@ -645,6 +657,7 @@ fn check_config(
     let StreamConfig {
         channels,
         sample_rate,
+        buffer_size,
     } = config;
     // Try and set the sample rate to what the user selected.
     let sample_rate = sample_rate.0.into();

src/host/coreaudio/mod.rs

@@ -5,7 +5,8 @@ use self::core_foundation_sys::string::{CFStringGetCString, CFStringGetCStringPt
 use self::coreaudio::audio_unit::render_callback::{self, data};
 use self::coreaudio::audio_unit::{AudioUnit, Element, Scope};
 use self::coreaudio::sys::{
-    kAudioDevicePropertyAvailableNominalSampleRates, kAudioDevicePropertyDeviceNameCFString,
+    kAudioDevicePropertyAvailableNominalSampleRates, kAudioDevicePropertyBufferFrameSize,
+    kAudioDevicePropertyBufferFrameSizeRange, kAudioDevicePropertyDeviceNameCFString,
     kAudioDevicePropertyNominalSampleRate, kAudioDevicePropertyScopeOutput,
     kAudioDevicePropertyStreamConfiguration, kAudioDevicePropertyStreamFormat,
     kAudioFormatFlagIsFloat, kAudioFormatFlagIsPacked, kAudioFormatLinearPCM,
@@ -20,10 +21,11 @@ use self::coreaudio::sys::{
 };
 use crate::traits::{DeviceTrait, HostTrait, StreamTrait};
 use crate::{
-    BackendSpecificError, BuildStreamError, ChannelCount, Data, DefaultStreamConfigError,
+    BackendSpecificError, BufferSize, BuildStreamError, ChannelCount, Data,
-    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
+    DefaultStreamConfigError, DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo,
-    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
+    PauseStreamError, PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError,
-    SupportedStreamConfigRange, SupportedStreamConfigsError,
+    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
+    SupportedStreamConfigsError,
 };
 use std::cell::RefCell;
 use std::ffi::CStr;
@@ -276,6 +278,9 @@ impl Device {
             let ranges: *mut AudioValueRange = ranges.as_mut_ptr() as *mut _;
             let ranges: &'static [AudioValueRange] = slice::from_raw_parts(ranges, n_ranges);
 
+            let audio_unit = audio_unit_from_device(self, true)?;
+            let buffer_size = get_io_buffer_frame_size_range(&audio_unit)?;
+
             // Collect the supported formats for the device.
             let mut fmts = vec![];
             for range in ranges {
@@ -283,6 +288,7 @@ impl Device {
                     channels: n_channels as ChannelCount,
                     min_sample_rate: SampleRate(range.mMinimum as _),
                     max_sample_rate: SampleRate(range.mMaximum as _),
+                    buffer_size: buffer_size.clone(),
                     sample_format: sample_format,
                 };
                 fmts.push(fmt);
@@ -374,9 +380,13 @@ impl Device {
                 }
             };
 
+            let audio_unit = audio_unit_from_device(self, true)?;
+            let buffer_size = get_io_buffer_frame_size_range(&audio_unit)?;
+
             let config = SupportedStreamConfig {
                 sample_rate: SampleRate(asbd.mSampleRate as _),
                 channels: asbd.mChannelsPerFrame as _,
+                buffer_size: buffer_size,
                 sample_format: sample_format,
             };
             Ok(config)
@@ -426,6 +436,24 @@ impl From<coreaudio::Error> for BuildStreamError {
     }
 }
 
+impl From<coreaudio::Error> for SupportedStreamConfigsError {
+    fn from(err: coreaudio::Error) -> SupportedStreamConfigsError {
+        let description = format!("{}", err);
+        let err = BackendSpecificError { description };
+        // Check for possible DeviceNotAvailable variant
+        SupportedStreamConfigsError::BackendSpecific { err }
+    }
+}
+
+impl From<coreaudio::Error> for DefaultStreamConfigError {
+    fn from(err: coreaudio::Error) -> DefaultStreamConfigError {
+        let description = format!("{}", err);
+        let err = BackendSpecificError { description };
+        // Check for possible DeviceNotAvailable variant
+        DefaultStreamConfigError::BackendSpecific { err }
+    }
+}
+
 // Create a coreaudio AudioStreamBasicDescription from a CPAL Format.
 fn asbd_from_config(
     config: &StreamConfig,
@@ -656,6 +684,29 @@ impl Device {
         let asbd = asbd_from_config(config, sample_format);
         audio_unit.set_property(kAudioUnitProperty_StreamFormat, scope, element, Some(&asbd))?;
 
+        // Set the buffersize
+        match config.buffer_size {
+            BufferSize::Fixed(v) => {
+                let buffer_size_range = get_io_buffer_frame_size_range(&audio_unit)?;
+                match buffer_size_range {
+                    SupportedBufferSize::Range { min, max } => {
+                        if v >= min && v <= max {
+                            audio_unit.set_property(
+                                kAudioDevicePropertyBufferFrameSize,
+                                scope,
+                                element,
+                                Some(&v),
+                            )?
+                        } else {
+                            return Err(BuildStreamError::StreamConfigNotSupported);
+                        }
+                    }
+                    SupportedBufferSize::Unknown => (),
+                }
+            }
+            BufferSize::Default => (),
+        }
+
         // Register the callback that is being called by coreaudio whenever it needs data to be
         // fed to the audio buffer.
         let bytes_per_channel = sample_format.sample_size();
@@ -727,6 +778,29 @@ impl Device {
         let asbd = asbd_from_config(config, sample_format);
         audio_unit.set_property(kAudioUnitProperty_StreamFormat, scope, element, Some(&asbd))?;
 
+        // Set the buffersize
+        match config.buffer_size {
+            BufferSize::Fixed(v) => {
+                let buffer_size_range = get_io_buffer_frame_size_range(&audio_unit)?;
+                match buffer_size_range {
+                    SupportedBufferSize::Range { min, max } => {
+                        if v >= min && v <= max {
+                            audio_unit.set_property(
+                                kAudioDevicePropertyBufferFrameSize,
+                                scope,
+                                element,
+                                Some(&v),
+                            )?
+                        } else {
+                            return Err(BuildStreamError::StreamConfigNotSupported);
+                        }
+                    }
+                    SupportedBufferSize::Unknown => (),
+                }
+            }
+            BufferSize::Default => (),
+        }
+
         // Register the callback that is being called by coreaudio whenever it needs data to be
         // fed to the audio buffer.
         let bytes_per_channel = sample_format.sample_size();
@@ -848,3 +922,18 @@ fn check_os_status(os_status: OSStatus) -> Result<(), BackendSpecificError> {
         }
     }
 }
+
+fn get_io_buffer_frame_size_range(
+    audio_unit: &AudioUnit,
+) -> Result<SupportedBufferSize, coreaudio::Error> {
+    let buffer_size_range: AudioValueRange = audio_unit.get_property(
+        kAudioDevicePropertyBufferFrameSizeRange,
+        Scope::Global,
+        Element::Output,
+    )?;
+
+    Ok(SupportedBufferSize::Range {
+        min: buffer_size_range.mMinimum as u32,
+        max: buffer_size_range.mMaximum as u32,
+    })
+}

src/host/emscripten/mod.rs

@@ -8,10 +8,10 @@ use stdweb::web::TypedArray;
 use stdweb::Reference;
 
 use crate::{
-    BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
+    BufferSize, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
     InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleFormat,
-    StreamConfig, StreamError, SupportedStreamConfig, SupportedStreamConfigRange,
+    SampleRate, StreamConfig, StreamError, SupportedBufferSize, SupportedStreamConfig,
-    SupportedStreamConfigsError,
+    SupportedStreamConfigRange, SupportedStreamConfigsError,
 };
 use traits::{DeviceTrait, HostTrait, StreamTrait};
 
@@ -41,6 +41,16 @@ pub struct StreamId(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 MIN_BUFFER_SIZE: u32 = 1;
+const MAX_BUFFER_SIZE: u32 = std::u32::MAX;
+const DEFAULT_BUFFER_SIZE: usize = 2048;
+const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;
+
 impl Host {
     pub fn new() -> Result<Self, crate::HostUnavailable> {
         stdweb::initialize();
@@ -71,21 +81,20 @@ impl Device {
     fn supported_output_configs(
         &self,
     ) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
-        // TODO: right now cpal's API doesn't allow flexibility here
+        let buffer_size = SupportedBufferSize::Range {
-        //       "44100" and "2" (channels) have also been hard-coded in the rest of the code ; if
+            min: MIN_BUFFER_SIZE,
-        //       this ever becomes more flexible, don't forget to change that
+            max: MAX_BUFFER_SIZE,
-        //       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.
+        let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
-        //
+            .map(|channels| SupportedStreamConfigRange {
-        //       UPDATE: We can do this now. Might be best to use `crate::COMMON_SAMPLE_RATES` and
+                channels,
-        //       filter out those that lay outside the range specified above.
+                min_sample_rate: MIN_SAMPLE_RATE,
-        Ok(vec![SupportedStreamConfigRange {
+                max_sample_rate: MAX_SAMPLE_RATE,
-            channels: 2,
+                buffer_size: buffer_size.clone(),
-            min_sample_rate: ::SampleRate(44100),
+                sample_format: SUPPORTED_SAMPLE_FORMAT,
-            max_sample_rate: ::SampleRate(44100),
+            })
-            sample_format: ::SampleFormat::F32,
+            .collect();
-        }]
+        Ok(configs.into_iter())
-        .into_iter())
     }
 
     fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
@@ -93,12 +102,15 @@ impl Device {
     }
 
     fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
-        // TODO: because it is hard coded, see supported_output_configs.
+        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)
     }
 }
 
@@ -169,7 +181,7 @@ impl DeviceTrait for Device {
 
     fn build_output_stream_raw<D, E>(
         &self,
-        _config: &StreamConfig,
+        config: &StreamConfig,
         sample_format: SampleFormat,
         data_callback: D,
         error_callback: E,
@@ -178,11 +190,20 @@ impl DeviceTrait for Device {
         D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
         E: FnMut(StreamError) + Send + 'static,
     {
-        assert_eq!(
+        if !valid_config(config, sample_format) {
-            sample_format,
+            return Err(BuildStreamError::StreamConfigNotSupported);
-            SampleFormat::F32,
+        }
-            "emscripten backend currently only supports `f32` data",
+
-        );
+        let buffer_size_frames = match config.buffer_size {
+            BufferSize::Fixed(v) => {
+                if v == 0 {
+                    return Err(BuildStreamError::StreamConfigNotSupported);
+                } else {
+                    v as usize
+                }
+            }
+            BufferSize::Default => DEFAULT_BUFFER_SIZE,
+        };
 
         // Create the stream.
         let audio_ctxt_ref = js!(return new AudioContext()).into_reference().unwrap();
@@ -199,7 +220,14 @@ impl DeviceTrait for Device {
         // See also: The call to `set_timeout` at the end of the `audio_callback_fn` which creates
         // the loop.
         set_timeout(
-            || audio_callback_fn::<D, E>(user_data_ptr as *mut c_void),
+            || {
+                audio_callback_fn::<D, E>(
+                    user_data_ptr as *mut c_void,
+                    config,
+                    sample_format,
+                    buffer_size_frames,
+                )
+            },
             10,
         );
 
@@ -223,12 +251,18 @@ impl StreamTrait for Stream {
 
 // The first argument of the callback function (a `void*`) is a casted pointer to `self`
 // and to the `callback` parameter that was passed to `run`.
-fn audio_callback_fn<D, E>(user_data_ptr: *mut c_void)
+fn audio_callback_fn<D, E>(
-where
+    user_data_ptr: *mut c_void,
+    config: &StreamConfig,
+    sample_format: SampleFormat,
+    buffer_size_frames: usize,
+) where
     D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
     E: FnMut(StreamError) + Send + 'static,
 {
-    const SAMPLE_RATE: usize = 44100;
+    let num_channels = config.channels as usize;
+    let sample_rate = config.sample_rate.0;
+    let buffer_size_samples = buffer_size_frames * num_channels;
 
     unsafe {
         let user_data_ptr2 = user_data_ptr as *mut (&Stream, D, E);
@@ -237,12 +271,11 @@ where
         let audio_ctxt = &stream.audio_ctxt_ref;
 
         // TODO: We should be re-using a buffer.
-        let mut temporary_buffer = vec![0.0; SAMPLE_RATE * 2 / 3];
+        let mut temporary_buffer = vec![0f32; buffer_size_samples];
 
         {
             let len = temporary_buffer.len();
             let data = temporary_buffer.as_mut_ptr() as *mut ();
-            let sample_format = SampleFormat::F32;
             let mut data = Data::from_parts(data, len, sample_format);
 
             let now_secs: f64 = js!(@{audio_ctxt}.getOutputTimestamp().currentTime)
@@ -253,7 +286,7 @@ where
             // we estimate based on buffer size instead. Probably should use this, but it's only
             // supported by firefox (2020-04-28).
             // let latency_secs: f64 = js!(@{audio_ctxt}.outputLatency).try_into().unwrap();
-            let buffer_duration = frames_to_duration(len, SAMPLE_RATE);
+            let buffer_duration = frames_to_duration(len, sample_rate as usize);
             let playback = callback
                 .add(buffer_duration)
                 .expect("`playback` occurs beyond representation supported by `StreamInstant`");
@@ -273,19 +306,19 @@ where
             typed_array
         };
 
-        let num_channels = 2u32; // TODO: correct value
         debug_assert_eq!(temporary_buffer.len() % num_channels as usize, 0);
 
         js!(
             var src_buffer = new Float32Array(@{typed_array}.buffer);
             var context = @{audio_ctxt};
-            var buf_len = @{temporary_buffer.len() as u32};
+            var buffer_size_frames = @{buffer_size_frames as u32};
-            var num_channels = @{num_channels};
+            var num_channels = @{num_channels as u32};
+            var sample_rate = sample_rate;
 
-            var buffer = context.createBuffer(num_channels, buf_len / num_channels, 44100);
+            var buffer = context.createBuffer(num_channels, buffer_size_frames, sample_rate);
             for (var channel = 0; channel < num_channels; ++channel) {
                 var buffer_content = buffer.getChannelData(channel);
-                for (var i = 0; i < buf_len / num_channels; ++i) {
+                for (var i = 0; i < buffer_size_frames; ++i) {
                     buffer_content[i] = src_buffer[i * num_channels + channel];
                 }
             }
@@ -299,7 +332,10 @@ where
         // TODO: handle latency better ; right now we just use setInterval with the amount of sound
         // data that is in each buffer ; this is obviously bad, and also the schedule is too tight
         // and there may be underflows
-        set_timeout(|| audio_callback_fn::<D, E>(user_data_ptr), 330);
+        set_timeout(
+            || audio_callback_fn::<D, E>(user_data_ptr, config, sample_format, buffer_size_frames),
+            buffer_size_frames as u32 * 1000 / sample_rate,
+        );
     }
 }
 
@@ -348,6 +384,15 @@ fn is_webaudio_available() -> bool {
     .unwrap()
 }
 
+// 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
+}
+
 // Convert the given duration in frames at the given sample rate to a `std::time::Duration`.
 fn frames_to_duration(frames: usize, rate: usize) -> std::time::Duration {
     let secsf = frames as f64 / rate as f64;

src/host/wasapi/device.rs

@@ -1,8 +1,8 @@
 use crate::{
-    BackendSpecificError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
+    BackendSpecificError, BufferSize, Data, DefaultStreamConfigError, DeviceNameError,
-    InputCallbackInfo, OutputCallbackInfo, SampleFormat, SampleRate, StreamConfig,
+    DevicesError, InputCallbackInfo, OutputCallbackInfo, SampleFormat, SampleRate, StreamConfig,
-    SupportedStreamConfig, SupportedStreamConfigRange, SupportedStreamConfigsError,
+    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
-    COMMON_SAMPLE_RATES,
+    SupportedStreamConfigsError, COMMON_SAMPLE_RATES,
 };
 use std;
 use std::ffi::OsString;
@@ -27,6 +27,7 @@ use super::winapi::shared::mmreg;
 use super::winapi::shared::winerror;
 use super::winapi::shared::wtypes;
 use super::winapi::Interface;
+
 // https://msdn.microsoft.com/en-us/library/cc230355.aspx
 use super::winapi::um::audioclient::{
     self, IAudioClient, IID_IAudioClient, AUDCLNT_E_DEVICE_INVALIDATED,
@@ -318,9 +319,11 @@ unsafe fn format_from_waveformatex_ptr(
         // Unknown data format returned by GetMixFormat.
         _ => return None,
     };
+
     let format = SupportedStreamConfig {
         channels: (*waveformatex_ptr).nChannels as _,
         sample_rate: SampleRate((*waveformatex_ptr).nSamplesPerSec),
+        buffer_size: SupportedBufferSize::Unknown,
         sample_format,
     };
     Some(format)
@@ -513,7 +516,7 @@ impl Device {
             // TODO: Test the different sample formats?
 
             // Create the supported formats.
-            let mut format = match format_from_waveformatex_ptr(default_waveformatex_ptr.0) {
+            let format = match format_from_waveformatex_ptr(default_waveformatex_ptr.0) {
                 Some(fmt) => fmt,
                 None => {
                     let description =
@@ -525,8 +528,13 @@ impl Device {
             };
             let mut supported_formats = Vec::with_capacity(supported_sample_rates.len());
             for rate in supported_sample_rates {
-                format.sample_rate = SampleRate(rate as _);
+                supported_formats.push(SupportedStreamConfigRange {
-                supported_formats.push(SupportedStreamConfigRange::from(format.clone()));
+                    channels: format.channels.clone(),
+                    min_sample_rate: SampleRate(rate as _),
+                    max_sample_rate: SampleRate(rate as _),
+                    buffer_size: format.buffer_size.clone(),
+                    sample_format: format.sample_format.clone(),
+                })
             }
             Ok(supported_formats.into_iter())
         }
@@ -639,6 +647,16 @@ impl Device {
                 }
             };
 
+            match config.buffer_size {
+                BufferSize::Fixed(_) => {
+                    // TO DO: We need IAudioClient3 to get buffersize ranges first
+                    // Otherwise the supported ranges are unknown. In the mean time
+                    // the smallest buffersize is selected and used.
+                    return Err(BuildStreamError::StreamConfigNotSupported);
+                }
+                BufferSize::Default => (),
+            };
+
             // Computing the format and initializing the device.
             let waveformatex = {
                 let format_attempt = config_to_waveformatextensible(config, sample_format)
@@ -791,6 +809,16 @@ impl Device {
                 }
             };
 
+            match config.buffer_size {
+                BufferSize::Fixed(_) => {
+                    // TO DO: We need IAudioClient3 to get buffersize ranges first
+                    // Otherwise the supported ranges are unknown. In the mean time
+                    // the smallest buffersize is selected and used.
+                    return Err(BuildStreamError::StreamConfigNotSupported);
+                }
+                BufferSize::Default => (),
+            };
+
             // Computing the format and initializing the device.
             let waveformatex = {
                 let format_attempt = config_to_waveformatextensible(config, sample_format)
@@ -813,6 +841,7 @@ impl Device {
                     &format_attempt.Format,
                     ptr::null(),
                 );
+
                 match check_result(hresult) {
                     Err(ref e) if e.raw_os_error() == Some(AUDCLNT_E_DEVICE_INVALIDATED) => {
                         (*audio_client).Release();

src/host/webaudio/mod.rs

@@ -7,10 +7,10 @@ use self::wasm_bindgen::prelude::*;
 use self::wasm_bindgen::JsCast;
 use self::web_sys::{AudioContext, AudioContextOptions};
 use crate::{
-    BackendSpecificError, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError,
+    BackendSpecificError, BufferSize, BuildStreamError, Data, DefaultStreamConfigError,
-    DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError,
+    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
-    SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
+    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedBufferSize,
-    SupportedStreamConfigRange, SupportedStreamConfigsError,
+    SupportedStreamConfig, SupportedStreamConfigRange, SupportedStreamConfigsError,
 };
 use std::ops::DerefMut;
 use std::sync::{Arc, Mutex, RwLock};
@@ -39,6 +39,9 @@ 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 MIN_BUFFER_SIZE: u32 = 1;
+const MAX_BUFFER_SIZE: u32 = std::u32::MAX;
+const DEFAULT_BUFFER_SIZE: usize = 2048;
 const SUPPORTED_SAMPLE_FORMAT: SampleFormat = SampleFormat::F32;
 
 impl Host {
@@ -93,11 +96,16 @@ impl Device {
     fn supported_output_configs(
         &self,
     ) -> Result<SupportedOutputConfigs, SupportedStreamConfigsError> {
+        let buffer_size = SupportedBufferSize::Range {
+            min: MIN_BUFFER_SIZE,
+            max: MAX_BUFFER_SIZE,
+        };
         let configs: Vec<_> = (MIN_CHANNELS..=MAX_CHANNELS)
             .map(|channels| SupportedStreamConfigRange {
                 channels,
                 min_sample_rate: MIN_SAMPLE_RATE,
                 max_sample_rate: MAX_SAMPLE_RATE,
+                buffer_size: buffer_size.clone(),
                 sample_format: SUPPORTED_SAMPLE_FORMAT,
             })
             .collect();
@@ -117,7 +125,7 @@ impl Device {
         configs.sort_by(|a, b| a.cmp_default_heuristics(b));
         let config = configs
             .into_iter()
-            .next()
+            .last()
             .expect(EXPECT)
             .with_sample_rate(DEFAULT_SAMPLE_RATE);
         Ok(config)
@@ -190,11 +198,20 @@ impl DeviceTrait for Device {
         }
 
         let n_channels = config.channels as usize;
-        // Use a buffer period of 1/3s for this early proof of concept.
+
-        // TODO: Change this to the requested buffer size when updating for the buffer size API.
+        let buffer_size_frames = match config.buffer_size {
-        let buffer_size_frames = (config.sample_rate.0 as f64 / 3.0).round() as usize;
+            BufferSize::Fixed(v) => {
+                if v == 0 {
+                    return Err(BuildStreamError::StreamConfigNotSupported);
+                } else {
+                    v as usize
+                }
+            }
+            BufferSize::Default => DEFAULT_BUFFER_SIZE,
+        };
         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.

src/lib.rs

@@ -180,6 +180,22 @@ pub type ChannelCount = u16;
 #[derive(Debug, Clone, Copy, PartialEq, Eq, PartialOrd, Ord)]
 pub struct SampleRate(pub u32);
 
+/// The desired number of frames for the hardware buffer.
+pub type FrameCount = u32;
+
+/// The buffer size used by the device.
+///
+/// Default is used when no specific buffer size is set and uses the default
+/// behavior of the given host. Note, the default buffer size may be surprisingly
+/// large, leading to latency issues. If low latency is desired, Fixed(BufferSize)
+/// should be used in accordance with the SupportedBufferSize range produced by
+/// the SupportedStreamConfig API.  
+#[derive(Clone, Debug, Eq, PartialEq)]
+pub enum BufferSize {
+    Default,
+    Fixed(FrameCount),
+}
+
 /// The set of parameters used to describe how to open a stream.
 ///
 /// The sample format is omitted in favour of using a sample type.
@@ -187,6 +203,19 @@ pub struct SampleRate(pub u32);
 pub struct StreamConfig {
     pub channels: ChannelCount,
     pub sample_rate: SampleRate,
+    pub buffer_size: BufferSize,
+}
+
+/// Describes the minimum and maximum supported buffer size for the device
+#[derive(Clone, Debug, Eq, PartialEq)]
+pub enum SupportedBufferSize {
+    Range {
+        min: FrameCount,
+        max: FrameCount,
+    },
+    /// In the case that the platform provides no way of getting the default
+    /// buffersize before starting a stream.
+    Unknown,
 }
 
 /// Describes a range of supported stream configurations, retrieved via the
@@ -198,6 +227,8 @@ pub struct SupportedStreamConfigRange {
     pub(crate) min_sample_rate: SampleRate,
     /// Maximum value for the samples rate of the supported formats.
     pub(crate) max_sample_rate: SampleRate,
+    /// Buffersize ranges supported by the device
+    pub(crate) buffer_size: SupportedBufferSize,
     /// Type of data expected by the device.
     pub(crate) sample_format: SampleFormat,
 }
@@ -208,6 +239,7 @@ pub struct SupportedStreamConfigRange {
 pub struct SupportedStreamConfig {
     channels: ChannelCount,
     sample_rate: SampleRate,
+    buffer_size: SupportedBufferSize,
     sample_format: SampleFormat,
 }
 
@@ -289,6 +321,10 @@ impl SupportedStreamConfig {
         self.sample_rate
     }
 
+    pub fn buffer_size(&self) -> &SupportedBufferSize {
+        &self.buffer_size
+    }
+
     pub fn sample_format(&self) -> SampleFormat {
         self.sample_format
     }
@@ -297,6 +333,7 @@ impl SupportedStreamConfig {
         StreamConfig {
             channels: self.channels,
             sample_rate: self.sample_rate,
+            buffer_size: BufferSize::Default,
         }
     }
 }
@@ -492,11 +529,15 @@ impl SupportedStreamConfigRange {
         self.max_sample_rate
     }
 
+    pub fn buffer_size(&self) -> &SupportedBufferSize {
+        &self.buffer_size
+    }
+
     pub fn sample_format(&self) -> SampleFormat {
         self.sample_format
     }
 
-    /// Retrieve a `SupportedStreamConfig` with the given sample rate.
+    /// Retrieve a `SupportedStreamConfig` with the given sample rate and buffer size.
     ///
     /// **panic!**s if the given `sample_rate` is outside the range specified within this
     /// `SupportedStreamConfigRange` instance.
@@ -504,8 +545,9 @@ impl SupportedStreamConfigRange {
         assert!(self.min_sample_rate <= sample_rate && sample_rate <= self.max_sample_rate);
         SupportedStreamConfig {
             channels: self.channels,
-            sample_format: self.sample_format,
             sample_rate,
+            sample_format: self.sample_format,
+            buffer_size: self.buffer_size,
         }
     }
 
@@ -516,6 +558,7 @@ impl SupportedStreamConfigRange {
             channels: self.channels,
             sample_rate: self.max_sample_rate,
             sample_format: self.sample_format,
+            buffer_size: self.buffer_size,
         }
     }
 
@@ -594,30 +637,35 @@ impl SupportedStreamConfigRange {
 fn test_cmp_default_heuristics() {
     let mut formats = vec![
         SupportedStreamConfigRange {
+            buffer_size: SupportedBufferSize::Range { min: 256, max: 512 },
             channels: 2,
             min_sample_rate: SampleRate(1),
             max_sample_rate: SampleRate(96000),
             sample_format: SampleFormat::F32,
         },
         SupportedStreamConfigRange {
+            buffer_size: SupportedBufferSize::Range { min: 256, max: 512 },
             channels: 1,
             min_sample_rate: SampleRate(1),
             max_sample_rate: SampleRate(96000),
             sample_format: SampleFormat::F32,
         },
         SupportedStreamConfigRange {
+            buffer_size: SupportedBufferSize::Range { min: 256, max: 512 },
             channels: 2,
             min_sample_rate: SampleRate(1),
             max_sample_rate: SampleRate(96000),
             sample_format: SampleFormat::I16,
         },
         SupportedStreamConfigRange {
+            buffer_size: SupportedBufferSize::Range { min: 256, max: 512 },
             channels: 2,
             min_sample_rate: SampleRate(1),
             max_sample_rate: SampleRate(96000),
             sample_format: SampleFormat::U16,
         },
         SupportedStreamConfigRange {
+            buffer_size: SupportedBufferSize::Range { min: 256, max: 512 },
             channels: 2,
             min_sample_rate: SampleRate(1),
             max_sample_rate: SampleRate(22050),
@@ -660,18 +708,6 @@ impl From<SupportedStreamConfig> for StreamConfig {
     }
 }
 
-impl From<SupportedStreamConfig> for SupportedStreamConfigRange {
-    #[inline]
-    fn from(format: SupportedStreamConfig) -> SupportedStreamConfigRange {
-        SupportedStreamConfigRange {
-            channels: format.channels,
-            min_sample_rate: format.sample_rate,
-            max_sample_rate: format.sample_rate,
-            sample_format: format.sample_format,
-        }
-    }
-}
-
 // If a backend does not provide an API for retrieving supported formats, we query it with a bunch
 // of commonly used rates. This is always the case for wasapi and is sometimes the case for alsa.
 //

src/platform/mod.rs

@@ -1,7 +1,7 @@
 //! Platform-specific items.
 //!
 //! This module also contains the implementation of the platform's dynamically dispatched `Host`
-//! type and its associated `EventLoop`, `Device`, `StreamId` and other associated types. These
+//! type and its associated `Device`, `StreamId` and other associated types. These
 //! types are useful in the case that users require switching between audio host APIs at runtime.
 
 #[doc(inline)]
@@ -27,7 +27,7 @@ pub use self::platform_impl::*;
 // }
 // ```
 //
-// And so on for Device, Devices, EventLoop, Host, StreamId, SupportedInputConfigs,
+// And so on for Device, Devices, Host, StreamId, SupportedInputConfigs,
 // SupportedOutputConfigs and all their necessary trait implementations.
 // ```
 macro_rules! impl_platform_host {
@@ -558,8 +558,8 @@ mod platform_impl {
 )))]
 mod platform_impl {
     pub use crate::host::null::{
-        Device as NullDevice, Devices as NullDevices, EventLoop as NullEventLoop, Host as NullHost,
+        Device as NullDevice, Devices as NullDevices, Host as NullHost,
-        StreamId as NullStreamId, SupportedInputConfigs as NullSupportedInputConfigs,
+        SupportedInputConfigs as NullSupportedInputConfigs,
         SupportedOutputConfigs as NullSupportedOutputConfigs,
     };