fix: stream config validation for all backends (#1215)

Author: roderickvd

CHANGELOG.md

@@ -72,9 +72,10 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
   silently returning an empty list.
 - **AAudio**: Bump MSRV to 1.85.
 - **AAudio**: Buffers with default sizes are now dynamically tuned.
-- **AAudio**: `SupportedBufferSize` now reports `min: 1`.
+- **AAudio**: `SupportedBufferSize` in enumeration is now `Unknown`.
 - **AAudio**: `default_input_config()` and `default_output_config()` now prefer 48 kHz, then
   44.1 kHz, then the maximum supported sample rate, instead of always taking the maximum.
+- **AAudio**: Channel enumeration extended to 8 channels.
 - **ALSA**: Stream error callback now receives `ErrorKind::DeviceNotAvailable` on device
   disconnection.
 - **ALSA**: Polling errors trigger underrun recovery instead of looping.
@@ -99,6 +100,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - **AudioWorklet**: `BufferSize::Fixed` now sets `renderSizeHint` on the `AudioContext`.
 - **AudioWorklet**: `default_output_config()` now uses 48 kHz as the default sample rate instead
   of 44.1 kHz, reflecting the dominant native rate on modern hardware.
+- **AudioWorklet**: `channels: 0` or `sample_rate: 0` now return `InvalidInput` instead of `UnsupportedConfig`.
+- **AudioWorklet**: Sample rates now enumerated as discrete standard rates in the spec-required
+  range of 3–768 kHz.
 - **CoreAudio**: Bump MSRV to 1.85.
 - **CoreAudio**: Bump `mach2` to 0.6 (uses `core::ffi` instead of `libc`, enables tvOS builds).
 - **CoreAudio**: Timestamps now include device latency and safety offset.
@@ -141,6 +145,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - **WebAudio**: Initial buffer scheduling offset now scales with buffer duration.
 - **WebAudio**: `default_output_config()` now uses 48 kHz as the default sample rate instead of
   44.1 kHz, reflecting the dominant native rate on modern hardware.
+- **WebAudio**: Sample rates now enumerated as discrete standard rates in the spec-required
+  range of 3–768 kHz.
 
 ### Removed
 
@@ -154,13 +160,17 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 ### Fixed
 
 - Fix numeric overflows in calls to create `StreamInstant` in ASIO, CoreAudio and JACK.
+- **AAudio**: Fix panic in device configuration enumeration for pathological channel counts.
 - **AAudio**: Fix thread lock when a stream is dropped before it fully starts.
 - **AAudio**: Fix capture and playback timestamps falling back to time-zero on error.
 - **AAudio**: Fix capture and playback timestamp not accounting for audio pipeline buffer depth.
 - **AAudio**: Fix overflow in `buffer_capacity_in_frames` for large fixed buffer sizes.
 - **AAudio**: Poisoned stream locks now return `ErrorKind::StreamInvalidated` instead of panicking.
 - **AAudio**: Output buffers are now zero-filled before the callback runs.
 - **AAudio**: Stream errors are now forwarded to `error_callback`.
+- **AAudio**: Fix `channels: 0` returning `UnsupportedConfig` instead of `InvalidInput`.
+- **AAudio**: Fix `sample_rate: 0` silently opening a stream at the NDK default rate instead of
+  returning `InvalidInput`.
 - **ALSA**: Fix capture stream hanging or spinning on overruns.
 - **ALSA**: Fix timestamps stepping backward during stream startup or after xrun recovery.
 - **ALSA**: Fix spurious timestamp errors during stream startup.
@@ -177,6 +187,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - **ALSA**: Fix `supported_configs()` using the same buffer range for all formats and channels.
 - **ALSA**: Fix `supported_configs()` dropping sample rates outside of `COMMON_SAMPLE_RATES`.
 - **ALSA**: Fix `BufferSize::Fixed(0)` being silently accepted.
+- **ALSA**: Fix `channels: 0` or `sample_rate: 0` returning `UnsupportedConfig` instead of `InvalidInput`.
+- **ALSA**: Fix `build_*_stream_raw` returning `UnsupportedConfig` instead of `UnsupportedOperation` when
+  the device does not support the requested direction.
 - **ASIO**: Fix enumeration returning only the first device when using `collect()`.
 - **ASIO**: Fix device enumeration and stream creation failing when called from spawned threads.
 - **ASIO**: Fix buffer size not resizing when the driver reports `kAsioBufferSizeChange`.
@@ -190,7 +203,17 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - **ASIO**: Fix `driver.sample_rate()` failures at stream creation being silently ignored.
 - **ASIO**: Fix callbacks firing before `build_*_stream` returns the `Stream` handle.
 - **ASIO**: Fix overrun not being reported when the driver reports `kAsioOverload`.
+- **ASIO**: Fix `channels: 0`, `sample_rate: 0`, or `BufferSize::Fixed(0)` not returning
+  `ErrorKind::InvalidInput`; preventing a divide-by-zero panic.
+- **ASIO**: Fix `BufferSize::Fixed` with a size that does not align to the driver's step constraint
+  not returning `ErrorKind::UnsupportedConfig`.
 - **AudioWorklet**: Fix `default_output_device()` to return `None` when AudioWorklet is unavailable.
+- **AudioWorklet**: Fix channel count exceeding `destination.maxChannelCount` silently using fewer
+  channels than requested.
+- **AudioWorklet**: Fix `supported_output_configs()` reporting the buffer size upper bound as
+  `FrameCount::MAX`; now correctly `floor(6 × sample_rate)` per spec.
+- **AudioWorklet**: Fix `supported_output_configs()` reporting the minimum render quantum size as
+  128 when `renderQuantumSize` is supported; the spec minimum is 1.
 - **CoreAudio**: Fix default output streams silently stopping when the system default output
   device is unplugged; they now reroute automatically or report `ErrorKind::DeviceNotAvailable`.
 - **CoreAudio**: Fix undefined behaviour and silent failure in loopback device creation.
@@ -201,23 +224,43 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - **CoreAudio**: Fix crashes on certain drivers due to early initialization.
 - **CoreAudio**: Fix `supported_output_configs()` and `supported_input_configs()` collapsing
   non-continuous hardware rates into a continuous range of sample rates (regression since v0.17.0).
+- **CoreAudio**: Fix `channels: 0`, `sample_rate: 0`, or `BufferSize::Fixed(0)` to return `ErrorKind::InvalidInput`.
+- **CoreAudio**: Fix `BufferSize::Fixed` producing cryptic backend errors when not validated against
+  the hardware buffer frame size range before stream creation.
+- **CoreAudio (iOS)**: Fix `BufferSize::Fixed` not being validated against the supported range before stream creation.
 - **JACK**: Fix input capture timestamp using callback execution time instead of cycle start.
 - **JACK**: Poisoned error callback mutex no longer silently drops subsequent error notifications.
 - **JACK**: Port registration failure now fails stream creation instead of silently failing.
 - **JACK**: `activate_async()` failure now returns an error instead of panicking.
 - **JACK**: Sample rate is now validated against the live JACK server at stream creation time.
 - **JACK**: Underrun notification no longer blocks the notification thread.
 - **JACK**: Output buffers are now zero-filled before the callback runs.
+- **JACK**: Fix `channels: 0`, `sample_rate: 0`, or `BufferSize::Fixed(0)` not returning `ErrorKind::InvalidInput`
+  before attempting server connection.
+- **JACK**: Fix `supported_input_configs()` and `supported_output_configs()` reporting a hardcoded sparse channel
+  list instead of enumerating all counts up to the number of physical system ports.
+- **PipeWire**: Fix `channels: 0` or `sample_rate: 0` silently using PipeWire-negotiated values instead of
+  returning `ErrorKind::InvalidInput`.
+- **PulseAudio**: Fix `channels: 0` or `sample_rate: 0` reaching the server instead of returning `ErrorKind::InvalidInput`.
 - **WASAPI**: Poisoned locks now returns an error instead of panicking.
 - **WASAPI**: Output buffers are now zero-filled before the callback runs.
 - **WASAPI**: Fix audio worker thread spawn failure panicking instead of returning an error.
 - **WASAPI**: Fix callbacks firing before `build_*_stream` returns the `Stream` handle.
 - **WASAPI**: Fix Communications-class inputs to return silence.
 - **WASAPI**: Fix `supported_input_configs()` advertising unsupported sample rates on input 
   devices.
+- **WASAPI**: Fix `sample_rate: 0` with `BufferSize::Fixed` causing a divide-by-zero panic.
+- **WASAPI**: Fix `channels: 0` or `sample_rate: 0` not returning `ErrorKind::InvalidInput`.
+- **WASAPI**: Fix `supported_input_configs()`, `supported_output_configs()`, `default_input_config()`,
+  and `default_output_config()` reporting an unconstrained buffer range on software audio stacks.
+- **PulseAudio**: Fix `supported_output_configs()` and `default_output_config()` to account for PulseAudio's double-buffer.
+- **WebAudio**: Fix overflow with pathological channel counts.
 - **WebAudio**: Fix duplicated callbacks on repeated `play()` calls.
 - **WebAudio**: Report errors through the callback instead of panicking.
 - **WebAudio**: Fix `default_output_device()` to return `None` when WebAudio is unavailable.
+- **WebAudio**: Fix `channels: 0`, `sample_rate: 0`, or `BufferSize::Fixed(0)` not returning `ErrorKind::InvalidInput`.
+- **WebAudio**: Fix channel count exceeding `destination.maxChannelCount` silently using fewer
+  channels than requested.
 
 
 ## [0.17.3] - 2026-02-18

asio-sys/CHANGELOG.md

@@ -8,7 +8,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 ## [Unreleased]
 
 ### Added
-- Added `Driver::latencies()`
+- Added `Driver::latencies()` to query input and output stream latencies in frames
+- Added `BufferPreference` enum expressing the driver's preferred buffer size and valid-size constraints
 - `asio_message` now dispatches `kAsioResyncRequest` and `kAsioLatenciesChanged` to callbacks
   instead of silently ignoring them
 - `sample_rate_did_change` now dispatches `AsioDriverEvent::SampleRateChanged` to registered
@@ -25,7 +26,7 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - Public-facing `c_long` fields and return types replaced with `i32`
 - Public-facing `c_double` parameters and return types replaced with `f64`
 - `Driver::latencies()` now returns `Latencies { input, output }`
-- `Driver::buffersize_range()` now returns `BufferSizeRange { min, max }`
+- `BufferSizeRange` adds `preferred: BufferPreference` field
 - `CallbackInfo::system_time` is now `u64` nanoseconds
 - `AsioError::ASE_NoMemory` renamed to `AsioError::NoMemory`
 - `AsioTime::reserved`, `AsioTimeInfo::reserved`, `AsioTimeCode::future` fields made private.

asio-sys/src/bindings/mod.rs

@@ -98,11 +98,20 @@ pub struct Latencies {
     pub output: i32,
 }
 
+/// Hardware buffer size preferences and constraints.
+#[derive(Clone, Copy, Debug, Eq, Hash, PartialEq)]
+pub enum BufferPreference {
+    Only(u32),
+    Preferred(u32),
+    Stepped { preferred: u32, step: u32 },
+}
+
 /// Minimum and maximum supported buffer sizes in frames.
 #[derive(Clone, Copy, Debug, Eq, Hash, PartialEq)]
 pub struct BufferSizeRange {
     pub min: i32,
     pub max: i32,
+    pub preferred: BufferPreference,
 }
 
 /// Information provided to the BufferCallback.
@@ -389,7 +398,7 @@ impl Asio {
         let mut driver_names: [[c_char; MAX_DRIVER_NAME_LEN]; MAX_DRIVERS] =
             [[0; MAX_DRIVER_NAME_LEN]; MAX_DRIVERS];
         // Pointer to each driver name.
-        let mut driver_name_ptrs: [*mut i8; MAX_DRIVERS] = [null_mut(); MAX_DRIVERS];
+        let mut driver_name_ptrs: [*mut c_char; MAX_DRIVERS] = [null_mut(); MAX_DRIVERS];
         for (ptr, name) in driver_name_ptrs.iter_mut().zip(&mut driver_names[..]) {
             *ptr = (*name).as_mut_ptr();
         }
@@ -450,7 +459,7 @@ impl Asio {
         let mut driver_info = std::mem::MaybeUninit::<ai::ASIODriverInfo>::uninit();
 
         unsafe {
-            match ai::load_asio_driver(driver_name_cstring.as_ptr() as *mut i8) {
+            match ai::load_asio_driver(driver_name_cstring.as_ptr() as *mut c_char) {
                 false => Err(LoadDriverError::LoadDriverFailed),
                 true => {
                     // Initialize ASIO.
@@ -527,6 +536,14 @@ impl Driver {
         Ok(BufferSizeRange {
             min: buffer_sizes.min,
             max: buffer_sizes.max,
+            preferred: match buffer_sizes.grans {
+                -1 => BufferPreference::Only(buffer_sizes.pref as u32),
+                0 => BufferPreference::Preferred(buffer_sizes.pref as u32),
+                granularity => BufferPreference::Stepped {
+                    preferred: buffer_sizes.pref as u32,
+                    step: granularity as u32,
+                },
+            },
         })
     }
 
@@ -600,7 +617,7 @@ impl Driver {
             let name_cstring = CString::new(self.inner.name.as_str())
                 .expect("driver name already stored must not contain null bytes");
             unsafe {
-                if !ai::load_asio_driver(name_cstring.as_ptr() as *mut i8) {
+                if !ai::load_asio_driver(name_cstring.as_ptr() as *mut c_char) {
                     return Err(AsioError::NoDrivers);
                 }
                 let mut driver_info = std::mem::MaybeUninit::<ai::ASIODriverInfo>::uninit();

src/host/aaudio/mod.rs

@@ -3,7 +3,6 @@
 //! Default backend on Android.
 
 use std::{
-    cmp,
     convert::TryInto,
     fmt,
     hash::{Hash, Hasher},
@@ -104,14 +103,9 @@ impl From<AndroidDeviceType> for InterfaceType {
     }
 }
 
-// constants from android.media.AudioFormat
-const CHANNEL_OUT_MONO: i32 = 4;
-const CHANNEL_OUT_STEREO: i32 = 12;
-
-// Android Java API supports up to 8 channels
-// TODO: more channels available in native AAudio
-// Maps channel masks to their corresponding channel counts
-const CHANNEL_CONFIGS: [(i32, ChannelCount); 2] = [(CHANNEL_OUT_MONO, 1), (CHANNEL_OUT_STEREO, 2)];
+// ITU-R BS.2051 standard surround channel counts; used as fallback when the device does not
+// report its own via AudioDeviceInfo.getChannelCounts().
+const DEFAULT_CHANNEL_COUNTS: [i32; 5] = [1, 2, 4, 6, 8];
 
 const SAMPLE_RATES: [i32; 15] = [
     5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000,
@@ -206,22 +200,22 @@ impl HostTrait for Host {
 }
 
 fn buffer_size_range() -> SupportedBufferSize {
-    SupportedBufferSize::Range {
-        min: 1,
-        max: i32::MAX as FrameCount,
-    }
+    // The valid range for frames_per_data_callback is any positive i32, but the meaningful
+    // lower bound (frames_per_burst) is only known after open_stream.
+    SupportedBufferSize::Unknown
 }
 
 fn default_supported_configs() -> VecIntoIter<SupportedStreamConfigRange> {
     const FORMATS: [SampleFormat; 2] = [SampleFormat::I16, SampleFormat::F32];
 
     let buffer_size = buffer_size_range();
-    let mut output = Vec::with_capacity(SAMPLE_RATES.len() * CHANNEL_CONFIGS.len() * FORMATS.len());
+    let mut output =
+        Vec::with_capacity(SAMPLE_RATES.len() * DEFAULT_CHANNEL_COUNTS.len() * FORMATS.len());
     for sample_format in &FORMATS {
-        for (_channel_mask, channel_count) in &CHANNEL_CONFIGS {
+        for channel_count in &DEFAULT_CHANNEL_COUNTS {
             for sample_rate in &SAMPLE_RATES {
                 output.push(SupportedStreamConfigRange {
-                    channels: *channel_count,
+                    channels: *channel_count as ChannelCount,
                     min_sample_rate: *sample_rate as SampleRate,
                     max_sample_rate: *sample_rate as SampleRate,
                     buffer_size,
@@ -241,11 +235,10 @@ fn device_supported_configs(device: &AudioDeviceInfo) -> VecIntoIter<SupportedSt
         &SAMPLE_RATES
     };
 
-    const ALL_CHANNELS: [i32; 2] = [1, 2];
     let channel_counts: &[i32] = if !device.channel_counts.is_empty() {
         &device.channel_counts
     } else {
-        &ALL_CHANNELS
+        &DEFAULT_CHANNEL_COUNTS
     };
 
     const ALL_FORMATS: [SampleFormat; 2] = [SampleFormat::I16, SampleFormat::F32];
@@ -259,15 +252,15 @@ fn device_supported_configs(device: &AudioDeviceInfo) -> VecIntoIter<SupportedSt
     let mut output = Vec::with_capacity(sample_rates.len() * channel_counts.len() * formats.len());
     for sample_rate in sample_rates {
         for channel_count in channel_counts {
-            assert!(*channel_count > 0);
-            if *channel_count > 2 {
-                // could be supported by the device
-                // TODO: more channels available in native AAudio
+            let Ok(channels) = ChannelCount::try_from(*channel_count) else {
+                continue;
+            };
+            if channels == 0 {
                 continue;
             }
             for format in formats {
                 output.push(SupportedStreamConfigRange {
-                    channels: cmp::min(*channel_count as ChannelCount, 2),
+                    channels,
                     min_sample_rate: *sample_rate as SampleRate,
                     max_sample_rate: *sample_rate as SampleRate,
                     buffer_size,
@@ -635,6 +628,7 @@ impl DeviceTrait for Device {
         D: FnMut(&Data, &InputCallbackInfo) + Send + 'static,
         E: FnMut(Error) + Send + 'static,
     {
+        crate::validate_stream_config(&config)?;
         let format = match sample_format {
             SampleFormat::I16 => ndk::audio::AudioFormat::PCM_I16,
             SampleFormat::F32 => ndk::audio::AudioFormat::PCM_Float,
@@ -645,21 +639,9 @@ impl DeviceTrait for Device {
                 ))
             }
         };
-        let channel_count = match config.channels {
-            1 => 1,
-            2 => 2,
-            channels => {
-                // TODO: more channels available in native AAudio
-                return Err(Error::with_message(
-                    ErrorKind::UnsupportedConfig,
-                    format!("Channel count {channels} is not supported"),
-                ));
-            }
-        };
-
         let builder = ndk::audio::AudioStreamBuilder::new()?
             .direction(ndk::audio::AudioDirection::Input)
-            .channel_count(channel_count)
+            .channel_count(config.channels as i32)
             .format(format);
 
         build_input_stream(
@@ -684,6 +666,7 @@ impl DeviceTrait for Device {
         D: FnMut(&mut Data, &OutputCallbackInfo) + Send + 'static,
         E: FnMut(Error) + Send + 'static,
     {
+        crate::validate_stream_config(&config)?;
         let format = match sample_format {
             SampleFormat::I16 => ndk::audio::AudioFormat::PCM_I16,
             SampleFormat::F32 => ndk::audio::AudioFormat::PCM_Float,
@@ -694,21 +677,9 @@ impl DeviceTrait for Device {
                 ))
             }
         };
-        let channel_count = match config.channels {
-            1 => 1,
-            2 => 2,
-            channels => {
-                // TODO: more channels available in native AAudio
-                return Err(Error::with_message(
-                    ErrorKind::UnsupportedConfig,
-                    format!("Channel count {channels} is not supported"),
-                ));
-            }
-        };
-
         let builder = ndk::audio::AudioStreamBuilder::new()?
             .direction(ndk::audio::AudioDirection::Output)
-            .channel_count(channel_count)
+            .channel_count(config.channels as i32)
             .format(format);
 
         build_output_stream(