Allow dead code for ALSA can_pause field

.github/workflows/cpal.yml

@@ -127,25 +127,6 @@ 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

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

Cargo.toml

@@ -1,6 +1,6 @@
 [package]
 name = "cpal"
-version = "0.12.1"
+version = "0.11.0"
 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"
@@ -9,10 +9,12 @@ license = "Apache-2.0"
 keywords = ["audio", "sound"]
 
 [features]
-asio = ["asio-sys", "num-traits"] # Only available on Windows. See README for setup instructions.
+asio = ["asio-sys"] # 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"
@@ -21,10 +23,8 @@ 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 = "0.2", path = "asio-sys", optional = true }
-num-traits = { version = "0.2.6", optional = true }
+asio-sys = { version = "0.1", path = "asio-sys", 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

@@ -26,10 +26,6 @@ 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 = "0.2.0"
+version = "0.1.0"
 authors = ["Tom Gowan <tomrgowan@gmail.com>"]
 description = "Low-level interface and binding generation for the steinberg ASIO SDK."
-repository = "https://github.com/RustAudio/cpal/"
+repository = "https://github.com/tomaka/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 = "0.54.0"
+bindgen = "0.51.0"
 walkdir = "2"
 cc = "1.0.25"
 
 [dependencies]
 lazy_static = "1.0.0"
-num-derive = "0.3"
+num-derive = "0.2"
 num-traits = "0.2"

asio-sys/src/bindings/errors.rs

@@ -19,7 +19,6 @@ pub enum AsioError {
     HardwareStuck,
     NoRate,
     ASE_NoMemory,
-    InvalidBufferSize,
     UnknownError,
 }
 
@@ -64,7 +63,6 @@ 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"),
         }
     }
@@ -96,7 +94,6 @@ 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

@@ -385,14 +385,6 @@ 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;
@@ -439,14 +431,8 @@ impl Driver {
     ///
     /// This will destroy any already allocated buffers.
     ///
-    /// If buffersize is None then the preferred buffer size from ASIO is used,
-    /// 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> {
+    /// The preferred buffer size from ASIO is used.
+    fn create_buffers(&self, buffer_infos: &mut [AsioBufferInfo]) -> Result<c_long, AsioError> {
         let num_channels = buffer_infos.len();
 
         // To pass as ai::ASIOCallbacks
@@ -463,17 +449,6 @@ 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()?;
@@ -485,27 +460,23 @@ impl Driver {
             asio_result!(ai::ASIOCreateBuffers(
                 buffer_infos.as_mut_ptr() as *mut _,
                 num_channels as i32,
-                buffer_size,
+                buffer_sizes.pref,
                 &mut callbacks as *mut _ as *mut _,
             ))?;
         }
         *state = DriverState::Prepared;
 
-        Ok(buffer_size)
+        Ok(buffer_sizes.pref)
     }
 
     /// 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(),
@@ -518,7 +489,7 @@ 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, buffer_size)?;
+                let buffer_size = self.create_buffers(&mut all_buffer_infos)?;
                 let output_buffer_infos = all_buffer_infos.split_off(split_point);
                 let input_buffer_infos = all_buffer_infos;
                 let input = Some(AsioStream {
@@ -533,7 +504,7 @@ impl Driver {
             }
             // Just input
             (false, true) => {
-                let buffer_size = self.create_buffers(&mut input_buffer_infos, buffer_size)?;
+                let buffer_size = self.create_buffers(&mut input_buffer_infos)?;
                 let input = Some(AsioStream {
                     buffer_infos: input_buffer_infos,
                     buffer_size,
@@ -543,7 +514,7 @@ impl Driver {
             }
             // Just output
             (true, false) => {
-                let buffer_size = self.create_buffers(&mut output_buffer_infos, buffer_size)?;
+                let buffer_size = self.create_buffers(&mut output_buffer_infos)?;
                 let input = None;
                 let output = Some(AsioStream {
                     buffer_infos: output_buffer_infos,
@@ -566,21 +537,17 @@ 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, buffer_size)
+        self.create_streams(input_buffer_infos, output_buffer_infos)
     }
 
     /// Prepare the output stream.
@@ -592,21 +559,17 @@ 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, buffer_size)
+        self.create_streams(input_buffer_infos, output_buffer_infos)
     }
 
     /// Releases buffers allocations.

examples/enumerate.rs

@@ -11,7 +11,6 @@ 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": "4.17.19",
-      "resolved": "https://registry.npmjs.org/lodash/-/lodash-4.17.19.tgz",
-      "integrity": "sha512-JNvd8XER9GQX0v2qJgsaN/mzFCNA5BRe/j8JN9d+tWyGLSodKQHKFicdwNYzWwI3wjRnaKPsGj1XkBjx/F96DQ=="
+      "version": "4.17.15",
+      "resolved": "https://registry.npmjs.org/lodash/-/lodash-4.17.15.tgz",
+      "integrity": "sha512-8xOcRHvCjnocdS5cpwXQXVzmmh5e5+saE2QGoeQmbKmRS6J3VQppPOIt0MnmE+4xlZoumy0GPG0D0MVIQbNA1A=="
     },
     "loglevel": {
       "version": "1.6.7",

src/host/alsa/mod.rs

@@ -3,17 +3,16 @@ extern crate libc;
 
 use self::alsa::poll::Descriptors;
 use crate::{
-    BackendSpecificError, BufferSize, BuildStreamError, ChannelCount, Data,
-    DefaultStreamConfigError, DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo,
-    PauseStreamError, PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError,
-    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
-    SupportedStreamConfigsError,
+    BackendSpecificError, BuildStreamError, ChannelCount, Data, DefaultStreamConfigError,
+    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
+    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, 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};
@@ -340,14 +339,6 @@ 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(),
         );
@@ -358,7 +349,6 @@ 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,
                     });
                 }
@@ -448,7 +438,6 @@ 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
@@ -881,7 +870,7 @@ fn set_hw_params_from_format<'a>(
     config: &StreamConfig,
     sample_format: SampleFormat,
 ) -> Result<alsa::pcm::HwParams<'a>, BackendSpecificError> {
-    let hw_params = alsa::pcm::HwParams::any(pcm_handle)?;
+    let mut hw_params = alsa::pcm::HwParams::any(pcm_handle)?;
     hw_params.set_access(alsa::pcm::Access::RWInterleaved)?;
 
     let sample_format = if cfg!(target_endian = "big") {
@@ -902,14 +891,11 @@ 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)?;
 
-    match config.buffer_size {
-        BufferSize::Fixed(v) => hw_params.set_buffer_size(v as alsa::pcm::Frames)?,
-        BufferSize::Default => {
-            // These values together represent a moderate latency and wakeup interval.
-            // 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)?;
-        }
+    // If this isn't set manually a overlarge buffer may be used causing audio delay
+    let mut hw_params_copy = hw_params.clone();
+    if let Err(_) = hw_params.set_buffer_time_near(100_000, alsa::ValueOr::Nearest) {
+        // Swap out the params with errors for a snapshot taken before the error was introduced.
+        mem::swap(&mut hw_params_copy, &mut hw_params);
     }
 
     pcm_handle.hw_params(&hw_params)?;

src/host/asio/device.rs

@@ -12,7 +12,6 @@ use DeviceNameError;
 use DevicesError;
 use SampleFormat;
 use SampleRate;
-use SupportedBufferSize;
 use SupportedStreamConfig;
 use SupportedStreamConfigRange;
 use SupportedStreamConfigsError;
@@ -78,13 +77,9 @@ impl Device {
                 continue;
             }
             for channels in 1..f.channels + 1 {
-                supported_configs.push(SupportedStreamConfigRange {
-                    channels,
-                    min_sample_rate: rate,
-                    max_sample_rate: rate,
-                    buffer_size: f.buffer_size.clone(),
-                    sample_format: f.sample_format.clone(),
-                })
+                f.channels = channels;
+                f.sample_rate = rate;
+                supported_configs.push(SupportedStreamConfigRange::from(f.clone()));
             }
         }
         Ok(supported_configs.into_iter())
@@ -115,13 +110,9 @@ impl Device {
                 continue;
             }
             for channels in 1..f.channels + 1 {
-                supported_configs.push(SupportedStreamConfigRange {
-                    channels,
-                    min_sample_rate: rate,
-                    max_sample_rate: rate,
-                    buffer_size: f.buffer_size.clone(),
-                    sample_format: f.sample_format.clone(),
-                })
+                f.channels = channels;
+                f.sample_rate = rate;
+                supported_configs.push(SupportedStreamConfigRange::from(f.clone()));
             }
         }
         Ok(supported_configs.into_iter())
@@ -131,11 +122,6 @@ 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)
@@ -143,7 +129,6 @@ impl Device {
         Ok(SupportedStreamConfig {
             channels,
             sample_rate,
-            buffer_size,
             sample_format,
         })
     }
@@ -152,18 +137,12 @@ 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, BufferSize, BuildStreamError, Data, InputCallbackInfo,
-    OutputCallbackInfo, PauseStreamError, PlayStreamError, Sample, SampleFormat, StreamConfig,
-    StreamError,
+    BackendSpecificError, BuildStreamError, Data, InputCallbackInfo, OutputCallbackInfo,
+    PauseStreamError, PlayStreamError, Sample, SampleFormat, StreamConfig, StreamError,
+    SupportedStreamConfig,
 };
 use std;
 use std::sync::atomic::{AtomicBool, Ordering};
@@ -482,12 +482,6 @@ 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 {
@@ -495,7 +489,7 @@ impl Device {
             None => {
                 let output = streams.output.take();
                 self.driver
-                    .prepare_input_stream(output, num_channels, buffer_size)
+                    .prepare_input_stream(output, num_channels)
                     .map(|new_streams| {
                         let bs = match new_streams.input {
                             Some(ref inp) => inp.buffer_size as usize,
@@ -529,12 +523,6 @@ 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 {
@@ -542,7 +530,7 @@ impl Device {
             None => {
                 let output = streams.output.take();
                 self.driver
-                    .prepare_output_stream(output, num_channels, buffer_size)
+                    .prepare_output_stream(output, num_channels)
                     .map(|new_streams| {
                         let bs = match new_streams.output {
                             Some(ref out) => out.buffer_size as usize,
@@ -657,7 +645,6 @@ 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

@@ -5,8 +5,7 @@ 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, kAudioDevicePropertyBufferFrameSize,
-    kAudioDevicePropertyBufferFrameSizeRange, kAudioDevicePropertyDeviceNameCFString,
+    kAudioDevicePropertyAvailableNominalSampleRates, kAudioDevicePropertyDeviceNameCFString,
     kAudioDevicePropertyNominalSampleRate, kAudioDevicePropertyScopeOutput,
     kAudioDevicePropertyStreamConfiguration, kAudioDevicePropertyStreamFormat,
     kAudioFormatFlagIsFloat, kAudioFormatFlagIsPacked, kAudioFormatLinearPCM,
@@ -21,11 +20,10 @@ use self::coreaudio::sys::{
 };
 use crate::traits::{DeviceTrait, HostTrait, StreamTrait};
 use crate::{
-    BackendSpecificError, BufferSize, BuildStreamError, ChannelCount, Data,
-    DefaultStreamConfigError, DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo,
-    PauseStreamError, PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError,
-    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
-    SupportedStreamConfigsError,
+    BackendSpecificError, BuildStreamError, ChannelCount, Data, DefaultStreamConfigError,
+    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
+    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
+    SupportedStreamConfigRange, SupportedStreamConfigsError,
 };
 use std::cell::RefCell;
 use std::ffi::CStr;
@@ -278,9 +276,6 @@ 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 {
@@ -288,7 +283,6 @@ 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);
@@ -380,13 +374,9 @@ 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)
@@ -436,24 +426,6 @@ 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,
@@ -684,29 +656,6 @@ 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();
@@ -778,29 +727,6 @@ 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();
@@ -922,18 +848,3 @@ 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::{
-    BufferSize, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
+    BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
     InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError, SampleFormat,
-    SampleRate, StreamConfig, StreamError, SupportedBufferSize, SupportedStreamConfig,
-    SupportedStreamConfigRange, SupportedStreamConfigsError,
+    StreamConfig, StreamError, SupportedStreamConfig, SupportedStreamConfigRange,
+    SupportedStreamConfigsError,
 };
 use traits::{DeviceTrait, HostTrait, StreamTrait};
 
@@ -41,16 +41,6 @@ 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();
@@ -81,20 +71,21 @@ 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();
-        Ok(configs.into_iter())
+        // 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())
     }
 
     fn default_input_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
@@ -102,15 +93,12 @@ impl Device {
     }
 
     fn default_output_config(&self) -> Result<SupportedStreamConfig, DefaultStreamConfigError> {
-        const EXPECT: &str = "expected at least one valid webaudio stream config";
-        let mut configs: Vec<_> = self.supported_output_configs().expect(EXPECT).collect();
-        configs.sort_by(|a, b| a.cmp_default_heuristics(b));
-        let config = configs
-            .into_iter()
-            .next()
-            .expect(EXPECT)
-            .with_sample_rate(DEFAULT_SAMPLE_RATE);
-        Ok(config)
+        // TODO: because it is hard coded, see supported_output_configs.
+        Ok(SupportedStreamConfig {
+            channels: 2,
+            sample_rate: ::SampleRate(44100),
+            sample_format: ::SampleFormat::F32,
+        })
     }
 }
 
@@ -181,7 +169,7 @@ 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,
@@ -190,20 +178,11 @@ impl DeviceTrait for Device {
         D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
         E: FnMut(StreamError) + Send + 'static,
     {
-        if !valid_config(config, sample_format) {
-            return Err(BuildStreamError::StreamConfigNotSupported);
-        }
-
-        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,
-        };
+        assert_eq!(
+            sample_format,
+            SampleFormat::F32,
+            "emscripten backend currently only supports `f32` data",
+        );
 
         // Create the stream.
         let audio_ctxt_ref = js!(return new AudioContext()).into_reference().unwrap();
@@ -220,14 +199,7 @@ 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,
-                    config,
-                    sample_format,
-                    buffer_size_frames,
-                )
-            },
+            || audio_callback_fn::<D, E>(user_data_ptr as *mut c_void),
             10,
         );
 
@@ -251,18 +223,12 @@ 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,
-    config: &StreamConfig,
-    sample_format: SampleFormat,
-    buffer_size_frames: usize,
-) where
+fn audio_callback_fn<D, E>(user_data_ptr: *mut c_void)
+where
     D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
     E: FnMut(StreamError) + Send + 'static,
 {
-    let num_channels = config.channels as usize;
-    let sample_rate = config.sample_rate.0;
-    let buffer_size_samples = buffer_size_frames * num_channels;
+    const SAMPLE_RATE: usize = 44100;
 
     unsafe {
         let user_data_ptr2 = user_data_ptr as *mut (&Stream, D, E);
@@ -271,11 +237,12 @@ fn audio_callback_fn<D, E>(
         let audio_ctxt = &stream.audio_ctxt_ref;
 
         // TODO: We should be re-using a buffer.
-        let mut temporary_buffer = vec![0f32; buffer_size_samples];
+        let mut temporary_buffer = vec![0.0; SAMPLE_RATE * 2 / 3];
 
         {
             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)
@@ -286,7 +253,7 @@ fn audio_callback_fn<D, E>(
             // 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 as usize);
+            let buffer_duration = frames_to_duration(len, SAMPLE_RATE);
             let playback = callback
                 .add(buffer_duration)
                 .expect("`playback` occurs beyond representation supported by `StreamInstant`");
@@ -306,19 +273,19 @@ fn audio_callback_fn<D, E>(
             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 buffer_size_frames = @{buffer_size_frames as u32};
-            var num_channels = @{num_channels as u32};
-            var sample_rate = sample_rate;
+            var buf_len = @{temporary_buffer.len() as u32};
+            var num_channels = @{num_channels};
 
-            var buffer = context.createBuffer(num_channels, buffer_size_frames, sample_rate);
+            var buffer = context.createBuffer(num_channels, buf_len / num_channels, 44100);
             for (var channel = 0; channel < num_channels; ++channel) {
                 var buffer_content = buffer.getChannelData(channel);
-                for (var i = 0; i < buffer_size_frames; ++i) {
+                for (var i = 0; i < buf_len / num_channels; ++i) {
                     buffer_content[i] = src_buffer[i * num_channels + channel];
                 }
             }
@@ -332,10 +299,7 @@ fn audio_callback_fn<D, E>(
         // 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, config, sample_format, buffer_size_frames),
-            buffer_size_frames as u32 * 1000 / sample_rate,
-        );
+        set_timeout(|| audio_callback_fn::<D, E>(user_data_ptr), 330);
     }
 }
 
@@ -384,15 +348,6 @@ 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, BufferSize, Data, DefaultStreamConfigError, DeviceNameError,
-    DevicesError, InputCallbackInfo, OutputCallbackInfo, SampleFormat, SampleRate, StreamConfig,
-    SupportedBufferSize, SupportedStreamConfig, SupportedStreamConfigRange,
-    SupportedStreamConfigsError, COMMON_SAMPLE_RATES,
+    BackendSpecificError, Data, DefaultStreamConfigError, DeviceNameError, DevicesError,
+    InputCallbackInfo, OutputCallbackInfo, SampleFormat, SampleRate, StreamConfig,
+    SupportedStreamConfig, SupportedStreamConfigRange, SupportedStreamConfigsError,
+    COMMON_SAMPLE_RATES,
 };
 use std;
 use std::ffi::OsString;
@@ -27,7 +27,6 @@ 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,
@@ -319,11 +318,9 @@ 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)
@@ -516,7 +513,7 @@ impl Device {
             // TODO: Test the different sample formats?
 
             // Create the supported formats.
-            let format = match format_from_waveformatex_ptr(default_waveformatex_ptr.0) {
+            let mut format = match format_from_waveformatex_ptr(default_waveformatex_ptr.0) {
                 Some(fmt) => fmt,
                 None => {
                     let description =
@@ -528,13 +525,8 @@ impl Device {
             };
             let mut supported_formats = Vec::with_capacity(supported_sample_rates.len());
             for rate in supported_sample_rates {
-                supported_formats.push(SupportedStreamConfigRange {
-                    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(),
-                })
+                format.sample_rate = SampleRate(rate as _);
+                supported_formats.push(SupportedStreamConfigRange::from(format.clone()));
             }
             Ok(supported_formats.into_iter())
         }
@@ -647,16 +639,6 @@ 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)
@@ -809,16 +791,6 @@ 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)
@@ -841,7 +813,6 @@ 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, BufferSize, BuildStreamError, Data, DefaultStreamConfigError,
-    DeviceNameError, DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError,
-    PlayStreamError, SampleFormat, SampleRate, StreamConfig, StreamError, SupportedBufferSize,
-    SupportedStreamConfig, SupportedStreamConfigRange, SupportedStreamConfigsError,
+    BackendSpecificError, BuildStreamError, Data, DefaultStreamConfigError, DeviceNameError,
+    DevicesError, InputCallbackInfo, OutputCallbackInfo, PauseStreamError, PlayStreamError,
+    SampleFormat, SampleRate, StreamConfig, StreamError, SupportedStreamConfig,
+    SupportedStreamConfigRange, SupportedStreamConfigsError,
 };
 use std::ops::DerefMut;
 use std::sync::{Arc, Mutex, RwLock};
@@ -39,9 +39,6 @@ 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 {
@@ -96,16 +93,11 @@ 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();
@@ -125,7 +117,7 @@ impl Device {
         configs.sort_by(|a, b| a.cmp_default_heuristics(b));
         let config = configs
             .into_iter()
-            .last()
+            .next()
             .expect(EXPECT)
             .with_sample_rate(DEFAULT_SAMPLE_RATE);
         Ok(config)
@@ -198,20 +190,11 @@ impl DeviceTrait for Device {
         }
 
         let n_channels = config.channels as usize;
-
-        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,
-        };
+        // 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 = (config.sample_rate.0 as f64 / 3.0).round() as usize;
         let buffer_size_samples = buffer_size_frames * n_channels;
         let buffer_time_step_secs = buffer_time_step_secs(buffer_size_frames, config.sample_rate);
-
         let data_callback = Arc::new(Mutex::new(Box::new(data_callback)));
 
         // Create the WebAudio stream.

src/lib.rs

@@ -180,22 +180,6 @@ 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.
@@ -203,19 +187,6 @@ pub enum BufferSize {
 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
@@ -227,8 +198,6 @@ 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,
 }
@@ -239,7 +208,6 @@ pub struct SupportedStreamConfigRange {
 pub struct SupportedStreamConfig {
     channels: ChannelCount,
     sample_rate: SampleRate,
-    buffer_size: SupportedBufferSize,
     sample_format: SampleFormat,
 }
 
@@ -321,10 +289,6 @@ impl SupportedStreamConfig {
         self.sample_rate
     }
 
-    pub fn buffer_size(&self) -> &SupportedBufferSize {
-        &self.buffer_size
-    }
-
     pub fn sample_format(&self) -> SampleFormat {
         self.sample_format
     }
@@ -333,7 +297,6 @@ impl SupportedStreamConfig {
         StreamConfig {
             channels: self.channels,
             sample_rate: self.sample_rate,
-            buffer_size: BufferSize::Default,
         }
     }
 }
@@ -529,15 +492,11 @@ 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 and buffer size.
+    /// Retrieve a `SupportedStreamConfig` with the given sample rate.
     ///
     /// **panic!**s if the given `sample_rate` is outside the range specified within this
     /// `SupportedStreamConfigRange` instance.
@@ -545,9 +504,8 @@ impl SupportedStreamConfigRange {
         assert!(self.min_sample_rate <= sample_rate && sample_rate <= self.max_sample_rate);
         SupportedStreamConfig {
             channels: self.channels,
-            sample_rate,
             sample_format: self.sample_format,
-            buffer_size: self.buffer_size,
+            sample_rate,
         }
     }
 
@@ -558,7 +516,6 @@ impl SupportedStreamConfigRange {
             channels: self.channels,
             sample_rate: self.max_sample_rate,
             sample_format: self.sample_format,
-            buffer_size: self.buffer_size,
         }
     }
 
@@ -637,35 +594,30 @@ 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),
@@ -708,6 +660,18 @@ 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 `Device`, `StreamId` and other associated types. These
+//! type and its associated `EventLoop`, `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, Host, StreamId, SupportedInputConfigs,
+// And so on for Device, Devices, EventLoop, 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, Host as NullHost,
-        SupportedInputConfigs as NullSupportedInputConfigs,
+        Device as NullDevice, Devices as NullDevices, EventLoop as NullEventLoop, Host as NullHost,
+        StreamId as NullStreamId, SupportedInputConfigs as NullSupportedInputConfigs,
         SupportedOutputConfigs as NullSupportedOutputConfigs,
     };