815 lines
31 KiB
Rust
815 lines
31 KiB
Rust
extern crate coreaudio;
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extern crate core_foundation_sys;
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use ChannelCount;
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use BackendSpecificError;
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use BuildStreamError;
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use DefaultFormatError;
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use DeviceNameError;
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use Format;
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use SupportedFormatsError;
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use Sample;
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use SampleFormat;
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use SampleRate;
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use StreamData;
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use SupportedFormat;
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use UnknownTypeInputBuffer;
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use UnknownTypeOutputBuffer;
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use std::ffi::CStr;
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use std::fmt;
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use std::mem;
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use std::os::raw::c_char;
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use std::ptr::null;
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use std::sync::{Arc, Mutex};
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use std::thread;
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use std::time::Duration;
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use std::slice;
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use self::coreaudio::audio_unit::{AudioUnit, Scope, Element};
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use self::coreaudio::audio_unit::render_callback::{self, data};
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use self::coreaudio::sys::{
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AudioBuffer,
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AudioBufferList,
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AudioDeviceID,
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AudioObjectAddPropertyListener,
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AudioObjectGetPropertyData,
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AudioObjectGetPropertyDataSize,
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AudioObjectID,
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AudioObjectPropertyAddress,
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AudioObjectPropertyScope,
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AudioObjectRemovePropertyListener,
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AudioObjectSetPropertyData,
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AudioStreamBasicDescription,
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AudioValueRange,
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kAudioDevicePropertyAvailableNominalSampleRates,
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kAudioDevicePropertyDeviceNameCFString,
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kAudioDevicePropertyNominalSampleRate,
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kAudioObjectPropertyScopeInput,
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kAudioObjectPropertyScopeGlobal,
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kAudioDevicePropertyScopeOutput,
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kAudioDevicePropertyStreamConfiguration,
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kAudioDevicePropertyStreamFormat,
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kAudioFormatFlagIsFloat,
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kAudioFormatFlagIsPacked,
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kAudioFormatLinearPCM,
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kAudioHardwareNoError,
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kAudioObjectPropertyElementMaster,
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kAudioObjectPropertyScopeOutput,
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kAudioOutputUnitProperty_CurrentDevice,
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kAudioOutputUnitProperty_EnableIO,
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kAudioUnitProperty_StreamFormat,
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kCFStringEncodingUTF8,
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OSStatus,
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};
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use self::core_foundation_sys::string::{
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CFStringRef,
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CFStringGetCStringPtr,
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};
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mod enumerate;
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pub use self::enumerate::{Devices, SupportedInputFormats, SupportedOutputFormats, default_input_device, default_output_device};
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#[derive(Clone, PartialEq, Eq)]
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pub struct Device {
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audio_device_id: AudioDeviceID,
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}
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impl Device {
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pub fn name(&self) -> Result<String, DeviceNameError> {
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let property_address = AudioObjectPropertyAddress {
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mSelector: kAudioDevicePropertyDeviceNameCFString,
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mScope: kAudioDevicePropertyScopeOutput,
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mElement: kAudioObjectPropertyElementMaster,
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};
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let device_name: CFStringRef = null();
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let data_size = mem::size_of::<CFStringRef>();
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let c_str = unsafe {
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let status = AudioObjectGetPropertyData(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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&device_name as *const _ as *mut _,
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);
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check_os_status(status)?;
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let c_string: *const c_char = CFStringGetCStringPtr(device_name, kCFStringEncodingUTF8);
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if c_string == null() {
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let description = "core foundation unexpectedly returned null string".to_string();
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let err = BackendSpecificError { description };
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return Err(err.into());
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}
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CStr::from_ptr(c_string as *mut _)
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};
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Ok(c_str.to_string_lossy().into_owned())
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}
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// Logic re-used between `supported_input_formats` and `supported_output_formats`.
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fn supported_formats(
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&self,
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scope: AudioObjectPropertyScope,
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) -> Result<SupportedOutputFormats, SupportedFormatsError>
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{
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let mut property_address = AudioObjectPropertyAddress {
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mSelector: kAudioDevicePropertyStreamConfiguration,
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mScope: scope,
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mElement: kAudioObjectPropertyElementMaster,
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};
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unsafe {
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// Retrieve the devices audio buffer list.
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let data_size = 0u32;
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let status = AudioObjectGetPropertyDataSize(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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);
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check_os_status(status)?;
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let mut audio_buffer_list: Vec<u8> = vec![];
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audio_buffer_list.reserve_exact(data_size as usize);
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let status = AudioObjectGetPropertyData(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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audio_buffer_list.as_mut_ptr() as *mut _,
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);
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check_os_status(status)?;
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let audio_buffer_list = audio_buffer_list.as_mut_ptr() as *mut AudioBufferList;
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// If there's no buffers, skip.
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if (*audio_buffer_list).mNumberBuffers == 0 {
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return Ok(vec![].into_iter());
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}
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// Count the number of channels as the sum of all channels in all output buffers.
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let n_buffers = (*audio_buffer_list).mNumberBuffers as usize;
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let first: *const AudioBuffer = (*audio_buffer_list).mBuffers.as_ptr();
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let buffers: &'static [AudioBuffer] = slice::from_raw_parts(first, n_buffers);
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let mut n_channels = 0;
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for buffer in buffers {
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n_channels += buffer.mNumberChannels as usize;
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}
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// AFAIK the sample format should always be f32 on macos and i16 on iOS? Feel free to
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// fix this if more pcm formats are supported.
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let sample_format = if cfg!(target_os = "ios") {
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SampleFormat::I16
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} else {
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SampleFormat::F32
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};
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// Get available sample rate ranges.
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property_address.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
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let data_size = 0u32;
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let status = AudioObjectGetPropertyDataSize(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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);
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check_os_status(status)?;
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let n_ranges = data_size as usize / mem::size_of::<AudioValueRange>();
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let mut ranges: Vec<u8> = vec![];
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ranges.reserve_exact(data_size as usize);
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let status = AudioObjectGetPropertyData(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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ranges.as_mut_ptr() as *mut _,
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);
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check_os_status(status)?;
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let ranges: *mut AudioValueRange = ranges.as_mut_ptr() as *mut _;
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let ranges: &'static [AudioValueRange] = slice::from_raw_parts(ranges, n_ranges);
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// Collect the supported formats for the device.
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let mut fmts = vec![];
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for range in ranges {
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let fmt = SupportedFormat {
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channels: n_channels as ChannelCount,
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min_sample_rate: SampleRate(range.mMinimum as _),
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max_sample_rate: SampleRate(range.mMaximum as _),
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data_type: sample_format,
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};
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fmts.push(fmt);
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}
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Ok(fmts.into_iter())
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}
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}
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pub fn supported_input_formats(&self) -> Result<SupportedOutputFormats, SupportedFormatsError> {
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self.supported_formats(kAudioObjectPropertyScopeInput)
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}
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pub fn supported_output_formats(&self) -> Result<SupportedOutputFormats, SupportedFormatsError> {
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self.supported_formats(kAudioObjectPropertyScopeOutput)
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}
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fn default_format(
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&self,
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scope: AudioObjectPropertyScope,
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) -> Result<Format, DefaultFormatError>
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{
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fn default_format_error_from_os_status(status: OSStatus) -> Result<(), DefaultFormatError> {
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let err = match coreaudio::Error::from_os_status(status) {
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Err(err) => err,
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Ok(_) => return Ok(()),
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};
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match err {
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coreaudio::Error::AudioUnit(coreaudio::error::AudioUnitError::FormatNotSupported) |
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coreaudio::Error::AudioCodec(_) |
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coreaudio::Error::AudioFormat(_) => {
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Err(DefaultFormatError::StreamTypeNotSupported)
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}
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coreaudio::Error::AudioUnit(coreaudio::error::AudioUnitError::NoConnection) => {
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Err(DefaultFormatError::DeviceNotAvailable)
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}
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err => {
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let description = format!("{}", std::error::Error::description(&err));
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let err = BackendSpecificError { description };
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Err(err.into())
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}
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}
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}
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let property_address = AudioObjectPropertyAddress {
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mSelector: kAudioDevicePropertyStreamFormat,
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mScope: scope,
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mElement: kAudioObjectPropertyElementMaster,
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};
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unsafe {
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let asbd: AudioStreamBasicDescription = mem::uninitialized();
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let data_size = mem::size_of::<AudioStreamBasicDescription>() as u32;
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let status = AudioObjectGetPropertyData(
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self.audio_device_id,
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&property_address as *const _,
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0,
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null(),
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&data_size as *const _ as *mut _,
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&asbd as *const _ as *mut _,
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);
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default_format_error_from_os_status(status)?;
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let sample_format = {
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let audio_format = coreaudio::audio_unit::AudioFormat::from_format_and_flag(
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asbd.mFormatID,
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Some(asbd.mFormatFlags),
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);
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let flags = match audio_format {
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Some(coreaudio::audio_unit::AudioFormat::LinearPCM(flags)) => flags,
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_ => return Err(DefaultFormatError::StreamTypeNotSupported),
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};
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let maybe_sample_format =
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coreaudio::audio_unit::SampleFormat::from_flags_and_bytes_per_frame(
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flags,
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asbd.mBytesPerFrame,
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);
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match maybe_sample_format {
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Some(coreaudio::audio_unit::SampleFormat::F32) => SampleFormat::F32,
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Some(coreaudio::audio_unit::SampleFormat::I16) => SampleFormat::I16,
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_ => return Err(DefaultFormatError::StreamTypeNotSupported),
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}
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};
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let format = Format {
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sample_rate: SampleRate(asbd.mSampleRate as _),
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channels: asbd.mChannelsPerFrame as _,
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data_type: sample_format,
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};
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Ok(format)
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}
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}
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pub fn default_input_format(&self) -> Result<Format, DefaultFormatError> {
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self.default_format(kAudioObjectPropertyScopeInput)
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}
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pub fn default_output_format(&self) -> Result<Format, DefaultFormatError> {
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self.default_format(kAudioObjectPropertyScopeOutput)
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}
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}
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impl fmt::Debug for Device {
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fn fmt(&self, f: &mut fmt::Formatter) -> fmt::Result {
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f.debug_struct("Device")
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.field("audio_device_id", &self.audio_device_id)
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.field("name", &self.name())
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.finish()
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}
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}
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// The ID of a stream is its index within the `streams` array of the events loop.
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#[derive(Debug, Clone, PartialEq, Eq, Hash)]
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pub struct StreamId(usize);
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pub struct EventLoop {
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// This `Arc` is shared with all the callbacks of coreaudio.
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active_callbacks: Arc<ActiveCallbacks>,
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streams: Mutex<Vec<Option<StreamInner>>>,
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}
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struct ActiveCallbacks {
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// Whenever the `run()` method is called with a callback, this callback is put in this list.
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callbacks: Mutex<Vec<&'static mut (FnMut(StreamId, StreamData) + Send)>>,
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}
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struct StreamInner {
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playing: bool,
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audio_unit: AudioUnit,
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// Track the device with which the audio unit was spawned.
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//
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// We must do this so that we can avoid changing the device sample rate if there is already
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// a stream associated with the device.
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device_id: AudioDeviceID,
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}
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// TODO need stronger error identification
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impl From<coreaudio::Error> for BuildStreamError {
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fn from(err: coreaudio::Error) -> BuildStreamError {
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match err {
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coreaudio::Error::RenderCallbackBufferFormatDoesNotMatchAudioUnitStreamFormat |
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coreaudio::Error::NoKnownSubtype |
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coreaudio::Error::AudioUnit(coreaudio::error::AudioUnitError::FormatNotSupported) |
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coreaudio::Error::AudioCodec(_) |
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coreaudio::Error::AudioFormat(_) => BuildStreamError::FormatNotSupported,
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_ => BuildStreamError::DeviceNotAvailable,
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}
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}
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}
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// Create a coreaudio AudioStreamBasicDescription from a CPAL Format.
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fn asbd_from_format(format: &Format) -> AudioStreamBasicDescription {
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let n_channels = format.channels as usize;
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let sample_rate = format.sample_rate.0;
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let bytes_per_channel = format.data_type.sample_size();
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let bits_per_channel = bytes_per_channel * 8;
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let bytes_per_frame = n_channels * bytes_per_channel;
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let frames_per_packet = 1;
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let bytes_per_packet = frames_per_packet * bytes_per_frame;
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let sample_format = format.data_type;
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let format_flags = match sample_format {
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SampleFormat::F32 => (kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked) as u32,
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_ => kAudioFormatFlagIsPacked as u32,
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};
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let asbd = AudioStreamBasicDescription {
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mBitsPerChannel: bits_per_channel as _,
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mBytesPerFrame: bytes_per_frame as _,
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mChannelsPerFrame: n_channels as _,
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mBytesPerPacket: bytes_per_packet as _,
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mFramesPerPacket: frames_per_packet as _,
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mFormatFlags: format_flags,
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mFormatID: kAudioFormatLinearPCM,
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mSampleRate: sample_rate as _,
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..Default::default()
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};
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asbd
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}
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fn audio_unit_from_device(device: &Device, input: bool) -> Result<AudioUnit, coreaudio::Error> {
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let mut audio_unit = {
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let au_type = if cfg!(target_os = "ios") {
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// The HalOutput unit isn't available in iOS unfortunately.
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// RemoteIO is a sensible replacement.
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// See https://goo.gl/CWwRTx
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coreaudio::audio_unit::IOType::RemoteIO
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} else {
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coreaudio::audio_unit::IOType::HalOutput
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};
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AudioUnit::new(au_type)?
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};
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if input {
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// Enable input processing.
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let enable_input = 1u32;
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audio_unit.set_property(
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kAudioOutputUnitProperty_EnableIO,
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Scope::Input,
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Element::Input,
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Some(&enable_input),
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)?;
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// Disable output processing.
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let disable_output = 0u32;
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audio_unit.set_property(
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kAudioOutputUnitProperty_EnableIO,
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Scope::Output,
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Element::Output,
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Some(&disable_output),
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)?;
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}
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audio_unit.set_property(
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kAudioOutputUnitProperty_CurrentDevice,
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Scope::Global,
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Element::Output,
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Some(&device.audio_device_id),
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)?;
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Ok(audio_unit)
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}
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impl EventLoop {
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#[inline]
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pub fn new() -> EventLoop {
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EventLoop {
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active_callbacks: Arc::new(ActiveCallbacks { callbacks: Mutex::new(Vec::new()) }),
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streams: Mutex::new(Vec::new()),
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}
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}
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|
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#[inline]
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pub fn run<F>(&self, mut callback: F) -> !
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where F: FnMut(StreamId, StreamData) + Send
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{
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{
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let callback: &mut (FnMut(StreamId, StreamData) + Send) = &mut callback;
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self.active_callbacks
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.callbacks
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.lock()
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.unwrap()
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.push(unsafe { mem::transmute(callback) });
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}
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loop {
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// So the loop does not get optimised out in --release
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thread::sleep(Duration::new(1u64, 0u32));
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}
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// Note: if we ever change this API so that `run` can return, then it is critical that
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// we remove the callback from `active_callbacks`.
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}
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fn next_stream_id(&self) -> usize {
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let streams_lock = self.streams.lock().unwrap();
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let stream_id = streams_lock
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.iter()
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.position(|n| n.is_none())
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.unwrap_or(streams_lock.len());
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stream_id
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}
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// Add the stream to the list of streams within `self`.
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fn add_stream(&self, stream_id: usize, au: AudioUnit, device_id: AudioDeviceID) {
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let inner = StreamInner {
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playing: true,
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audio_unit: au,
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device_id: device_id,
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};
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let mut streams_lock = self.streams.lock().unwrap();
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if stream_id == streams_lock.len() {
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streams_lock.push(Some(inner));
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} else {
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streams_lock[stream_id] = Some(inner);
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}
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}
|
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|
|
#[inline]
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|
pub fn build_input_stream(
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&self,
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device: &Device,
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format: &Format,
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) -> Result<StreamId, BuildStreamError>
|
|
{
|
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// The scope and element for working with a device's input stream.
|
|
let scope = Scope::Output;
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let element = Element::Input;
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|
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// Check whether or not we need to change the device sample rate to suit the one specified for the stream.
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unsafe {
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// Get the current sample rate.
|
|
let mut property_address = AudioObjectPropertyAddress {
|
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mSelector: kAudioDevicePropertyNominalSampleRate,
|
|
mScope: kAudioObjectPropertyScopeGlobal,
|
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mElement: kAudioObjectPropertyElementMaster,
|
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};
|
|
let sample_rate: f64 = 0.0;
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let data_size = mem::size_of::<f64>() as u32;
|
|
let status = AudioObjectGetPropertyData(
|
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device.audio_device_id,
|
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&property_address as *const _,
|
|
0,
|
|
null(),
|
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&data_size as *const _ as *mut _,
|
|
&sample_rate as *const _ as *mut _,
|
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);
|
|
coreaudio::Error::from_os_status(status)?;
|
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|
|
// If the requested sample rate is different to the device sample rate, update the device.
|
|
if sample_rate as u32 != format.sample_rate.0 {
|
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|
|
// In order to avoid breaking existing input streams we return an error if there is
|
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// already an active input stream for this device with the actual sample rate.
|
|
for stream in &*self.streams.lock().unwrap() {
|
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if let Some(stream) = stream.as_ref() {
|
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if stream.device_id == device.audio_device_id {
|
|
let description = "cannot change device sample rate for stream as an \
|
|
existing stream is already running at the current sample rate"
|
|
.into();
|
|
let err = BackendSpecificError { description };
|
|
return Err(err.into());
|
|
}
|
|
}
|
|
}
|
|
|
|
// Get available sample rate ranges.
|
|
property_address.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
|
|
let data_size = 0u32;
|
|
let status = AudioObjectGetPropertyDataSize(
|
|
device.audio_device_id,
|
|
&property_address as *const _,
|
|
0,
|
|
null(),
|
|
&data_size as *const _ as *mut _,
|
|
);
|
|
coreaudio::Error::from_os_status(status)?;
|
|
let n_ranges = data_size as usize / mem::size_of::<AudioValueRange>();
|
|
let mut ranges: Vec<u8> = vec![];
|
|
ranges.reserve_exact(data_size as usize);
|
|
let status = AudioObjectGetPropertyData(
|
|
device.audio_device_id,
|
|
&property_address as *const _,
|
|
0,
|
|
null(),
|
|
&data_size as *const _ as *mut _,
|
|
ranges.as_mut_ptr() as *mut _,
|
|
);
|
|
coreaudio::Error::from_os_status(status)?;
|
|
let ranges: *mut AudioValueRange = ranges.as_mut_ptr() as *mut _;
|
|
let ranges: &'static [AudioValueRange] = slice::from_raw_parts(ranges, n_ranges);
|
|
|
|
// Now that we have the available ranges, pick the one matching the desired rate.
|
|
let sample_rate = format.sample_rate.0;
|
|
let maybe_index = ranges
|
|
.iter()
|
|
.position(|r| r.mMinimum as u32 == sample_rate && r.mMaximum as u32 == sample_rate);
|
|
let range_index = match maybe_index {
|
|
None => return Err(BuildStreamError::FormatNotSupported),
|
|
Some(i) => i,
|
|
};
|
|
|
|
// Update the property selector to specify the nominal sample rate.
|
|
property_address.mSelector = kAudioDevicePropertyNominalSampleRate;
|
|
|
|
// Setting the sample rate of a device is an asynchronous process in coreaudio.
|
|
//
|
|
// Thus we are required to set a `listener` so that we may be notified when the
|
|
// change occurs.
|
|
unsafe extern "C" fn rate_listener(
|
|
device_id: AudioObjectID,
|
|
_n_addresses: u32,
|
|
_properties: *const AudioObjectPropertyAddress,
|
|
rate_ptr: *mut ::std::os::raw::c_void,
|
|
) -> OSStatus {
|
|
let rate_ptr: *const f64 = rate_ptr as *const _;
|
|
let data_size = mem::size_of::<f64>();
|
|
let property_address = AudioObjectPropertyAddress {
|
|
mSelector: kAudioDevicePropertyNominalSampleRate,
|
|
mScope: kAudioObjectPropertyScopeGlobal,
|
|
mElement: kAudioObjectPropertyElementMaster,
|
|
};
|
|
AudioObjectGetPropertyData(
|
|
device_id,
|
|
&property_address as *const _,
|
|
0,
|
|
null(),
|
|
&data_size as *const _ as *mut _,
|
|
rate_ptr as *const _ as *mut _,
|
|
)
|
|
}
|
|
|
|
// Add our sample rate change listener callback.
|
|
let reported_rate: f64 = 0.0;
|
|
let status = AudioObjectAddPropertyListener(
|
|
device.audio_device_id,
|
|
&property_address as *const _,
|
|
Some(rate_listener),
|
|
&reported_rate as *const _ as *mut _,
|
|
);
|
|
coreaudio::Error::from_os_status(status)?;
|
|
|
|
// Finally, set the sample rate.
|
|
let sample_rate = sample_rate as f64;
|
|
let status = AudioObjectSetPropertyData(
|
|
device.audio_device_id,
|
|
&property_address as *const _,
|
|
0,
|
|
null(),
|
|
data_size,
|
|
&ranges[range_index] as *const _ as *const _,
|
|
);
|
|
coreaudio::Error::from_os_status(status)?;
|
|
|
|
// Wait for the reported_rate to change.
|
|
//
|
|
// This should not take longer than a few ms, but we timeout after 1 sec just in case.
|
|
let timer = ::std::time::Instant::now();
|
|
while sample_rate != reported_rate {
|
|
if timer.elapsed() > ::std::time::Duration::from_secs(1) {
|
|
let description = "timeout waiting for sample rate update for device".into();
|
|
let err = BackendSpecificError { description };
|
|
return Err(err.into());
|
|
}
|
|
::std::thread::sleep(::std::time::Duration::from_millis(5));
|
|
}
|
|
|
|
// Remove the `rate_listener` callback.
|
|
let status = AudioObjectRemovePropertyListener(
|
|
device.audio_device_id,
|
|
&property_address as *const _,
|
|
Some(rate_listener),
|
|
&reported_rate as *const _ as *mut _,
|
|
);
|
|
coreaudio::Error::from_os_status(status)?;
|
|
}
|
|
}
|
|
|
|
let mut audio_unit = audio_unit_from_device(device, true)?;
|
|
|
|
// Set the stream in interleaved mode.
|
|
let asbd = asbd_from_format(format);
|
|
audio_unit.set_property(kAudioUnitProperty_StreamFormat, scope, element, Some(&asbd))?;
|
|
|
|
// Determine the future ID of the stream.
|
|
let stream_id = self.next_stream_id();
|
|
|
|
// Register the callback that is being called by coreaudio whenever it needs data to be
|
|
// fed to the audio buffer.
|
|
let active_callbacks = self.active_callbacks.clone();
|
|
let sample_format = format.data_type;
|
|
let bytes_per_channel = format.data_type.sample_size();
|
|
type Args = render_callback::Args<data::Raw>;
|
|
audio_unit.set_input_callback(move |args: Args| unsafe {
|
|
let ptr = (*args.data.data).mBuffers.as_ptr() as *const AudioBuffer;
|
|
let len = (*args.data.data).mNumberBuffers as usize;
|
|
let buffers: &[AudioBuffer] = slice::from_raw_parts(ptr, len);
|
|
|
|
// TODO: Perhaps loop over all buffers instead?
|
|
let AudioBuffer {
|
|
mNumberChannels: _num_channels,
|
|
mDataByteSize: data_byte_size,
|
|
mData: data
|
|
} = buffers[0];
|
|
|
|
let mut callbacks = active_callbacks.callbacks.lock().unwrap();
|
|
|
|
// A small macro to simplify handling the callback for different sample types.
|
|
macro_rules! try_callback {
|
|
($SampleFormat:ident, $SampleType:ty) => {{
|
|
let data_len = (data_byte_size as usize / bytes_per_channel) as usize;
|
|
let data_slice = slice::from_raw_parts(data as *const $SampleType, data_len);
|
|
let callback = match callbacks.get_mut(0) {
|
|
Some(cb) => cb,
|
|
None => return Ok(()),
|
|
};
|
|
let unknown_type_buffer = UnknownTypeInputBuffer::$SampleFormat(::InputBuffer { buffer: data_slice });
|
|
let stream_data = StreamData::Input { buffer: unknown_type_buffer };
|
|
callback(StreamId(stream_id), stream_data);
|
|
}};
|
|
}
|
|
|
|
match sample_format {
|
|
SampleFormat::F32 => try_callback!(F32, f32),
|
|
SampleFormat::I16 => try_callback!(I16, i16),
|
|
SampleFormat::U16 => try_callback!(U16, u16),
|
|
}
|
|
|
|
Ok(())
|
|
})?;
|
|
|
|
// TODO: start playing now? is that consistent with the other backends?
|
|
audio_unit.start()?;
|
|
|
|
// Add the stream to the list of streams within `self`.
|
|
self.add_stream(stream_id, audio_unit, device.audio_device_id);
|
|
|
|
Ok(StreamId(stream_id))
|
|
}
|
|
|
|
#[inline]
|
|
pub fn build_output_stream(
|
|
&self,
|
|
device: &Device,
|
|
format: &Format,
|
|
) -> Result<StreamId, BuildStreamError>
|
|
{
|
|
let mut audio_unit = audio_unit_from_device(device, false)?;
|
|
|
|
// The scope and element for working with a device's output stream.
|
|
let scope = Scope::Input;
|
|
let element = Element::Output;
|
|
|
|
// Set the stream in interleaved mode.
|
|
let asbd = asbd_from_format(format);
|
|
audio_unit.set_property(kAudioUnitProperty_StreamFormat, scope, element, Some(&asbd))?;
|
|
|
|
// Determine the future ID of the stream.
|
|
let stream_id = self.next_stream_id();
|
|
|
|
// Register the callback that is being called by coreaudio whenever it needs data to be
|
|
// fed to the audio buffer.
|
|
let active_callbacks = self.active_callbacks.clone();
|
|
let sample_format = format.data_type;
|
|
let bytes_per_channel = format.data_type.sample_size();
|
|
type Args = render_callback::Args<data::Raw>;
|
|
audio_unit.set_render_callback(move |args: Args| unsafe {
|
|
// If `run()` is currently running, then a callback will be available from this list.
|
|
// Otherwise, we just fill the buffer with zeroes and return.
|
|
|
|
let AudioBuffer {
|
|
mNumberChannels: _num_channels,
|
|
mDataByteSize: data_byte_size,
|
|
mData: data
|
|
} = (*args.data.data).mBuffers[0];
|
|
|
|
let mut callbacks = active_callbacks.callbacks.lock().unwrap();
|
|
|
|
// A small macro to simplify handling the callback for different sample types.
|
|
macro_rules! try_callback {
|
|
($SampleFormat:ident, $SampleType:ty, $equilibrium:expr) => {{
|
|
let data_len = (data_byte_size as usize / bytes_per_channel) as usize;
|
|
let data_slice = slice::from_raw_parts_mut(data as *mut $SampleType, data_len);
|
|
let callback = match callbacks.get_mut(0) {
|
|
Some(cb) => cb,
|
|
None => {
|
|
for sample in data_slice.iter_mut() {
|
|
*sample = $equilibrium;
|
|
}
|
|
return Ok(());
|
|
}
|
|
};
|
|
let unknown_type_buffer = UnknownTypeOutputBuffer::$SampleFormat(::OutputBuffer { buffer: data_slice });
|
|
let stream_data = StreamData::Output { buffer: unknown_type_buffer };
|
|
callback(StreamId(stream_id), stream_data);
|
|
}};
|
|
}
|
|
|
|
match sample_format {
|
|
SampleFormat::F32 => try_callback!(F32, f32, 0.0),
|
|
SampleFormat::I16 => try_callback!(I16, i16, 0),
|
|
SampleFormat::U16 => try_callback!(U16, u16, ::std::u16::MAX / 2),
|
|
}
|
|
|
|
Ok(())
|
|
})?;
|
|
|
|
// TODO: start playing now? is that consistent with the other backends?
|
|
audio_unit.start()?;
|
|
|
|
// Add the stream to the list of streams within `self`.
|
|
self.add_stream(stream_id, audio_unit, device.audio_device_id);
|
|
|
|
Ok(StreamId(stream_id))
|
|
}
|
|
|
|
pub fn destroy_stream(&self, stream_id: StreamId) {
|
|
let mut streams = self.streams.lock().unwrap();
|
|
streams[stream_id.0] = None;
|
|
}
|
|
|
|
pub fn play_stream(&self, stream: StreamId) {
|
|
let mut streams = self.streams.lock().unwrap();
|
|
let stream = streams[stream.0].as_mut().unwrap();
|
|
|
|
if !stream.playing {
|
|
stream.audio_unit.start().unwrap();
|
|
stream.playing = true;
|
|
}
|
|
}
|
|
|
|
pub fn pause_stream(&self, stream: StreamId) {
|
|
let mut streams = self.streams.lock().unwrap();
|
|
let stream = streams[stream.0].as_mut().unwrap();
|
|
|
|
if stream.playing {
|
|
stream.audio_unit.stop().unwrap();
|
|
stream.playing = false;
|
|
}
|
|
}
|
|
}
|
|
|
|
fn check_os_status(os_status: OSStatus) -> Result<(), BackendSpecificError> {
|
|
match coreaudio::Error::from_os_status(os_status) {
|
|
Ok(()) => Ok(()),
|
|
Err(err) => {
|
|
let description = std::error::Error::description(&err).to_string();
|
|
Err(BackendSpecificError { description })
|
|
}
|
|
}
|
|
}
|