Second round of correctness fixes on multiple backends

CHANGELOG.md

@@ -41,13 +41,23 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 ### Fixed
 
 - **ALSA**: Fix a remaining timestamp segfault on 32-bit platforms with a 64-bit kernel `time_t`.
+- **ALSA**: Improved enumeration accuracy for supported format, channel and rate combinations.
 - **ASIO**: Fix a deadlock when dropping a `Stream` that owns another ASIO `Stream`.
 - **ASIO**: Fix loading a driver while a previous driver was still unloading.
+- **ASIO**: `Stream` no longer risks blocking or panicking in the driver callback while another stream is being created or destroyed.
 - **AudioWorklet**: Fix processor construction failures not being reported to `error_callback`.
 - **AudioWorklet**: Fix dropouts in output streams when the callback buffer grows.
+- **CoreAudio**: Fix the device running at a different sample rate from the stream on hardware that reports a continuous rate range.
+- **CoreAudio**: Fix `supported_configs()` only reporting `F32`, even on hardware that also supports other sample formats.
 - **JACK**: Channel enumeration is capped at the physical system port count again.
+- **JACK**: Streams no longer panic when the server delivers a larger period than the negotiated buffer size.
+- **PipeWire**: Fix an empty chunk being emitted when a cycle requests no frames.
+- **PipeWire**: Fix capture reading from the wrong offset in the buffer on some devices.
 - **WASAPI**: Device enumeration no longer panics if the COM enumerator fails to initialize.
 - **WASAPI**: Revert "Default device changes no longer report `DeviceChanged`" as it was misinformed.
+- **WASAPI**: Output streams now start with real audio immediately instead of undefined content in the render buffer.
+- **WASAPI**: A stream paused immediately after starting no longer plays silence before real audio on resume.
+- **WASAPI**: Fix `I64` and `F64` incorrectly reported as supported output formats.
 
 ## [0.18.2] - 2026-08-16
 

src/host/alsa/mod.rs

@@ -535,71 +535,81 @@ impl Device {
             //SND_PCM_FORMAT_U18_3BE,
         ];
 
-        let min_rate = hw_params.get_rate_min()?;
-        let max_rate = hw_params.get_rate_max()?;
-
-        let sample_rates = if min_rate == max_rate || hw_params.test_rate(min_rate + 1).is_ok() {
-            // Fixed rate or continuous range.
-            vec![(min_rate, max_rate)]
-        } else {
-            // Discrete rates: probe the standard list plus the hardware's own min and max so
-            // that rates outside `COMMON_SAMPLE_RATES` are not missed.
-            let mut probe: Vec<SampleRate> = COMMON_SAMPLE_RATES.to_vec();
-            probe.push(min_rate);
-            probe.push(max_rate);
-            probe.sort_unstable();
-            probe.dedup();
-            probe
-                .into_iter()
-                .filter(|&r| (min_rate..=max_rate).contains(&r) && hw_params.test_rate(r).is_ok())
-                .map(|r| (r, r))
-                .collect()
-        };
-
-        let min_channels = hw_params.get_channels_min()?;
         // 64 = AES10 (MADI) maximum; also prevents spinning on plugins like plughw that report u32::MAX.
         const CHANNEL_ENUM_CAP: u32 = 64;
-        let max_channels = hw_params
-            .get_channels_max()?
-            .min(CHANNEL_ENUM_CAP)
-            .min(ChannelCount::MAX as u32);
-
-        let supported_channels: Vec<ChannelCount> =
-            if min_channels == max_channels || hw_params.test_channels(min_channels + 1).is_ok() {
-                (min_channels..=max_channels)
-                    .map(|c| c as ChannelCount)
-                    .collect()
-            } else {
-                (min_channels..=max_channels)
-                    .filter(|&c| hw_params.test_channels(c).is_ok())
-                    .map(|c| c as ChannelCount)
-                    .collect()
-            };
 
-        let mut output =
-            Vec::with_capacity(FORMATS.len() * supported_channels.len() * sample_rates.len());
+        let mut output = Vec::new();
         let mut seen_formats: Vec<SampleFormat> = Vec::with_capacity(FORMATS.len());
 
         // Key: (channels, physical width in bits) with 4 physical widths (8/16/32/64 bits)
         let mut buffer_size_cache: HashMap<(ChannelCount, u32), SupportedBufferSize> =
-            HashMap::with_capacity(supported_channels.len() * 4);
+            HashMap::new();
 
+        // `test_*` checks a value, it doesn't apply it, so format/channels/rate are each set on a
+        // clone in sequence rather than tested independently against the same unconstrained params.
         for &(sample_format, alsa_format) in FORMATS.iter() {
-            if seen_formats.contains(&sample_format) || hw_params.test_format(alsa_format).is_err()
-            {
+            if seen_formats.contains(&sample_format) {
+                continue;
+            }
+            let format_params = hw_params.clone();
+            if format_params.set_format(alsa_format).is_err() {
                 continue;
             }
             seen_formats.push(sample_format);
             let width = alsa_format.physical_width().unwrap_or(0) as u32;
 
-            for &channels in &supported_channels {
+            let (Ok(min_channels), Ok(max_channels)) = (
+                format_params.get_channels_min(),
+                format_params.get_channels_max(),
+            ) else {
+                continue;
+            };
+            let max_channels = max_channels
+                .min(CHANNEL_ENUM_CAP)
+                .min(ChannelCount::MAX as u32);
+
+            for raw_channels in min_channels..=max_channels {
+                let channel_params = format_params.clone();
+                if channel_params.set_channels(raw_channels).is_err() {
+                    continue;
+                }
+                let channels = raw_channels as ChannelCount;
+
                 let buffer_size =
                     *buffer_size_cache
                         .entry((channels, width))
                         .or_insert_with(|| {
                             supported_period_size_range(&hw_params, alsa_format, channels)
                         });
 
+                let (Ok(min_rate), Ok(max_rate)) =
+                    (channel_params.get_rate_min(), channel_params.get_rate_max())
+                else {
+                    continue;
+                };
+
+                let sample_rates =
+                    if min_rate == max_rate || channel_params.test_rate(min_rate + 1).is_ok() {
+                        // Fixed rate or continuous range.
+                        vec![(min_rate, max_rate)]
+                    } else {
+                        // Discrete rates: probe the standard list plus the hardware's own min and max
+                        // so that rates outside `COMMON_SAMPLE_RATES` are not missed.
+                        let mut probe: Vec<SampleRate> = COMMON_SAMPLE_RATES.to_vec();
+                        probe.push(min_rate);
+                        probe.push(max_rate);
+                        probe.sort_unstable();
+                        probe.dedup();
+                        probe
+                            .into_iter()
+                            .filter(|&r| {
+                                (min_rate..=max_rate).contains(&r)
+                                    && channel_params.test_rate(r).is_ok()
+                            })
+                            .map(|r| (r, r))
+                            .collect()
+                    };
+
                 for &(min_rate, max_rate) in sample_rates.iter() {
                     output.push(SupportedStreamConfigRange {
                         channels,

src/host/asio/stream.rs

@@ -223,8 +223,9 @@ impl Device {
             }
             last_buffer_index = callback_info.buffer_index;
 
-            // There is 0% chance of lock contention the host only locks when recreating streams.
-            let stream_lock = asio_streams.lock().unwrap();
+            let Ok(stream_lock) = asio_streams.lock() else {
+                return;
+            };
             let asio_stream = match stream_lock.input {
                 Some(ref asio_stream) => asio_stream,
                 None => return,
@@ -577,8 +578,9 @@ impl Device {
             }
             last_buffer_index = callback_info.buffer_index;
 
-            // There is 0% chance of lock contention the host only locks when recreating streams.
-            let mut stream_lock = asio_streams.lock().unwrap();
+            let Ok(mut stream_lock) = asio_streams.lock() else {
+                return;
+            };
             let asio_stream = match stream_lock.output {
                 Some(ref mut asio_stream) => asio_stream,
                 None => return,

src/host/coreaudio/macos/device.rs

@@ -15,7 +15,7 @@ use coreaudio::audio_unit::{
     audio_format::LinearPcmFlags,
     macos_helpers::{
         RateListener, audio_unit_from_device_id_uninitialized, find_matching_physical_format,
-        get_device_name, set_device_physical_stream_format,
+        get_device_name, get_supported_physical_stream_formats, set_device_physical_stream_format,
     },
     render_callback::{self, data},
 };
@@ -526,18 +526,7 @@ impl Device {
                 n_channels += buf.mNumberChannels as usize;
             }
 
-            // TODO: macOS should support U8, I16, I32, F32 and F64. This should allow for using
-            // I16 but just use F32 for now as it's the default anyway.
-            let sample_format = SampleFormat::F32;
-
             // Get available sample rate ranges.
-            // The property "kAudioDevicePropertyAvailableNominalSampleRates" returns a list of pairs of
-            // minimum and maximum sample rates but most of the devices returns pairs of same values though the underlying mechanism is unclear.
-            // This may cause issues when, for example, sorting the configs by the sample rates.
-            // We follows the implementation of RtAudio, which returns single element of config
-            // when all the pairs have the same values and returns multiple elements otherwise.
-            // See https://github.com/thestk/rtaudio/blob/master/RtAudio.cpp#L1369C1-L1375C39
-
             property_address.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
             let mut data_size = 0u32;
             let status = AudioObjectGetPropertyDataSize(
@@ -575,19 +564,55 @@ impl Device {
             }
             let buffer_size = get_io_buffer_frame_size_range(self.audio_device_id)?;
 
-            // Most hardware reports discrete rates (mMinimum == mMaximum); some aggregate or
-            // virtual devices report continuous ranges.
-            let fmts: Vec<_> = ranges
-                .iter()
-                .map(|range| SupportedStreamConfigRange {
-                    channels: n_channels as ChannelCount,
-                    min_sample_rate: range.mMinimum as u32,
-                    max_sample_rate: range.mMaximum as u32,
-                    buffer_size,
-                    sample_format,
-                })
-                .collect();
-            Ok(fmts.into_iter())
+            // AUHAL always converts to and from F32 regardless of the physical format, so
+            // advertise it at every nominal rate (most hardware reports discrete rates, i.e.
+            // mMinimum == mMaximum; some aggregate or virtual devices report continuous ranges).
+            let f32_fmts = ranges.iter().map(|range| SupportedStreamConfigRange {
+                channels: n_channels as ChannelCount,
+                min_sample_rate: range.mMinimum as u32,
+                max_sample_rate: range.mMaximum as u32,
+                buffer_size,
+                sample_format: SampleFormat::F32,
+            });
+
+            // The hardware's own physical formats, so integer-only devices advertise their
+            // bit-perfect paths instead of only the AUHAL-converted F32 one.
+            let physical_fmts = get_supported_physical_stream_formats(self.audio_device_id)
+                .unwrap_or_default()
+                .into_iter()
+                .filter_map(|fmt| {
+                    let Some(coreaudio::audio_unit::AudioFormat::LinearPCM(flags)) =
+                        coreaudio::audio_unit::AudioFormat::from_format_and_flag(
+                            fmt.mFormat.mFormatID,
+                            Some(fmt.mFormat.mFormatFlags),
+                        )
+                    else {
+                        return None;
+                    };
+                    let sample_format = match CoreAudioSampleFormat::from_flags_and_bits_per_sample(
+                        flags,
+                        fmt.mFormat.mBitsPerChannel,
+                    )? {
+                        CoreAudioSampleFormat::I8 => SampleFormat::I8,
+                        CoreAudioSampleFormat::I16 => SampleFormat::I16,
+                        CoreAudioSampleFormat::I24 => SampleFormat::I24,
+                        CoreAudioSampleFormat::I32 => SampleFormat::I32,
+                        // Already covered by f32_fmts at every rate, not just this row's range.
+                        CoreAudioSampleFormat::F32 => return None,
+                    };
+                    Some(SupportedStreamConfigRange {
+                        channels: fmt.mFormat.mChannelsPerFrame as ChannelCount,
+                        min_sample_rate: fmt.mSampleRateRange.mMinimum as u32,
+                        max_sample_rate: fmt.mSampleRateRange.mMaximum as u32,
+                        buffer_size,
+                        sample_format,
+                    })
+                });
+
+            Ok(f32_fmts
+                .chain(physical_fmts)
+                .collect::<Vec<_>>()
+                .into_iter())
         }
     }
 
@@ -711,14 +736,15 @@ impl Device {
 
         // Set the physical stream format (bit depth + sample rate) on the hardware device.
         // This avoids unnecessary format conversions, which is especially important on aggregate
-        // devices. Falls back to sample-rate-only if no matching physical format is available.
-        if set_physical_format(
+        // devices. Falls back to sample-rate-only if no matching physical format is available, or
+        // if the closest match found doesn't actually run at the requested rate.
+        if !set_physical_format(
             self.audio_device_id,
             config.sample_rate,
             config.channels,
             sample_format,
         )
-        .is_err()
+        .is_ok_and(|asbd| (asbd.mSampleRate - config.sample_rate as f64).abs() < 1.0)
         {
             set_sample_rate(self.audio_device_id, config.sample_rate, timeout)?;
         }
@@ -845,14 +871,15 @@ impl Device {
 
         // Best-effort: set the physical stream format (bit depth + sample rate) on the hardware.
         // This avoids unnecessary conversions, especially on aggregate devices. Not an error if
-        // it fails — the AudioUnit will handle format conversion as before.
-        if set_physical_format(
+        // it fails: the AudioUnit will handle format conversion as before. Also falls back if the
+        // closest match found doesn't actually run at the requested rate.
+        if !set_physical_format(
             self.audio_device_id,
             config.sample_rate,
             config.channels,
             sample_format,
         )
-        .is_err()
+        .is_ok_and(|asbd| (asbd.mSampleRate - config.sample_rate as f64).abs() < 1.0)
         {
             set_sample_rate(self.audio_device_id, config.sample_rate, timeout)?;
         }