extern crate coreaudio; extern crate core_foundation_sys; use ChannelCount; use BackendSpecificError; use BuildStreamError; use DefaultFormatError; use DeviceNameError; use Format; use SupportedFormatsError; use Sample; use SampleFormat; use SampleRate; use StreamData; use SupportedFormat; use UnknownTypeInputBuffer; use UnknownTypeOutputBuffer; use std::ffi::CStr; use std::fmt; use std::mem; use std::os::raw::c_char; use std::ptr::null; use std::sync::{Arc, Mutex}; use std::thread; use std::time::Duration; use std::slice; use self::coreaudio::audio_unit::{AudioUnit, Scope, Element}; use self::coreaudio::audio_unit::render_callback::{self, data}; use self::coreaudio::sys::{ AudioBuffer, AudioBufferList, AudioDeviceID, AudioObjectAddPropertyListener, AudioObjectGetPropertyData, AudioObjectGetPropertyDataSize, AudioObjectID, AudioObjectPropertyAddress, AudioObjectPropertyScope, AudioObjectRemovePropertyListener, AudioObjectSetPropertyData, AudioStreamBasicDescription, AudioValueRange, kAudioDevicePropertyAvailableNominalSampleRates, kAudioDevicePropertyDeviceNameCFString, kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeInput, kAudioObjectPropertyScopeGlobal, kAudioDevicePropertyScopeOutput, kAudioDevicePropertyStreamConfiguration, kAudioDevicePropertyStreamFormat, kAudioFormatFlagIsFloat, kAudioFormatFlagIsPacked, kAudioFormatLinearPCM, kAudioHardwareNoError, kAudioObjectPropertyElementMaster, kAudioObjectPropertyScopeOutput, kAudioOutputUnitProperty_CurrentDevice, kAudioOutputUnitProperty_EnableIO, kAudioUnitProperty_StreamFormat, kCFStringEncodingUTF8, OSStatus, }; use self::core_foundation_sys::string::{ CFStringRef, CFStringGetCStringPtr, }; mod enumerate; pub use self::enumerate::{Devices, SupportedInputFormats, SupportedOutputFormats, default_input_device, default_output_device}; #[derive(Clone, PartialEq, Eq)] pub struct Device { audio_device_id: AudioDeviceID, } impl Device { pub fn name(&self) -> Result { let property_address = AudioObjectPropertyAddress { mSelector: kAudioDevicePropertyDeviceNameCFString, mScope: kAudioDevicePropertyScopeOutput, mElement: kAudioObjectPropertyElementMaster, }; let device_name: CFStringRef = null(); let data_size = mem::size_of::(); let c_str = unsafe { let status = AudioObjectGetPropertyData( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, &device_name as *const _ as *mut _, ); check_os_status(status)?; let c_string: *const c_char = CFStringGetCStringPtr(device_name, kCFStringEncodingUTF8); if c_string == null() { let description = "core foundation unexpectedly returned null string".to_string(); let err = BackendSpecificError { description }; return Err(err.into()); } CStr::from_ptr(c_string as *mut _) }; Ok(c_str.to_string_lossy().into_owned()) } // Logic re-used between `supported_input_formats` and `supported_output_formats`. fn supported_formats( &self, scope: AudioObjectPropertyScope, ) -> Result { let mut property_address = AudioObjectPropertyAddress { mSelector: kAudioDevicePropertyStreamConfiguration, mScope: scope, mElement: kAudioObjectPropertyElementMaster, }; unsafe { // Retrieve the devices audio buffer list. let data_size = 0u32; let status = AudioObjectGetPropertyDataSize( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, ); check_os_status(status)?; let mut audio_buffer_list: Vec = vec![]; audio_buffer_list.reserve_exact(data_size as usize); let status = AudioObjectGetPropertyData( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, audio_buffer_list.as_mut_ptr() as *mut _, ); check_os_status(status)?; let audio_buffer_list = audio_buffer_list.as_mut_ptr() as *mut AudioBufferList; // If there's no buffers, skip. if (*audio_buffer_list).mNumberBuffers == 0 { return Ok(vec![].into_iter()); } // Count the number of channels as the sum of all channels in all output buffers. let n_buffers = (*audio_buffer_list).mNumberBuffers as usize; let first: *const AudioBuffer = (*audio_buffer_list).mBuffers.as_ptr(); let buffers: &'static [AudioBuffer] = slice::from_raw_parts(first, n_buffers); let mut n_channels = 0; for buffer in buffers { n_channels += buffer.mNumberChannels as usize; } // AFAIK the sample format should always be f32 on macos and i16 on iOS? Feel free to // fix this if more pcm formats are supported. let sample_format = if cfg!(target_os = "ios") { SampleFormat::I16 } else { SampleFormat::F32 }; // Get available sample rate ranges. property_address.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; let data_size = 0u32; let status = AudioObjectGetPropertyDataSize( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, ); check_os_status(status)?; let n_ranges = data_size as usize / mem::size_of::(); let mut ranges: Vec = vec![]; ranges.reserve_exact(data_size as usize); let status = AudioObjectGetPropertyData( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, ranges.as_mut_ptr() as *mut _, ); check_os_status(status)?; let ranges: *mut AudioValueRange = ranges.as_mut_ptr() as *mut _; let ranges: &'static [AudioValueRange] = slice::from_raw_parts(ranges, n_ranges); // Collect the supported formats for the device. let mut fmts = vec![]; for range in ranges { let fmt = SupportedFormat { channels: n_channels as ChannelCount, min_sample_rate: SampleRate(range.mMinimum as _), max_sample_rate: SampleRate(range.mMaximum as _), data_type: sample_format, }; fmts.push(fmt); } Ok(fmts.into_iter()) } } pub fn supported_input_formats(&self) -> Result { self.supported_formats(kAudioObjectPropertyScopeInput) } pub fn supported_output_formats(&self) -> Result { self.supported_formats(kAudioObjectPropertyScopeOutput) } fn default_format( &self, scope: AudioObjectPropertyScope, ) -> Result { fn default_format_error_from_os_status(status: OSStatus) -> Option { let err = match coreaudio::Error::from_os_status(status) { Err(err) => err, Ok(_) => return None, }; match err { coreaudio::Error::RenderCallbackBufferFormatDoesNotMatchAudioUnitStreamFormat | coreaudio::Error::NoKnownSubtype | coreaudio::Error::AudioUnit(coreaudio::error::AudioUnitError::FormatNotSupported) | coreaudio::Error::AudioCodec(_) | coreaudio::Error::AudioFormat(_) => Some(DefaultFormatError::StreamTypeNotSupported), _ => Some(DefaultFormatError::DeviceNotAvailable), } } let property_address = AudioObjectPropertyAddress { mSelector: kAudioDevicePropertyStreamFormat, mScope: scope, mElement: kAudioObjectPropertyElementMaster, }; unsafe { let asbd: AudioStreamBasicDescription = mem::uninitialized(); let data_size = mem::size_of::() as u32; let status = AudioObjectGetPropertyData( self.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, &asbd as *const _ as *mut _, ); if status != kAudioHardwareNoError as i32 { let err = default_format_error_from_os_status(status) .expect("no known error for OSStatus"); return Err(err); } let sample_format = { let audio_format = coreaudio::audio_unit::AudioFormat::from_format_and_flag( asbd.mFormatID, Some(asbd.mFormatFlags), ); let flags = match audio_format { Some(coreaudio::audio_unit::AudioFormat::LinearPCM(flags)) => flags, _ => return Err(DefaultFormatError::StreamTypeNotSupported), }; let maybe_sample_format = coreaudio::audio_unit::SampleFormat::from_flags_and_bytes_per_frame( flags, asbd.mBytesPerFrame, ); match maybe_sample_format { Some(coreaudio::audio_unit::SampleFormat::F32) => SampleFormat::F32, Some(coreaudio::audio_unit::SampleFormat::I16) => SampleFormat::I16, _ => return Err(DefaultFormatError::StreamTypeNotSupported), } }; let format = Format { sample_rate: SampleRate(asbd.mSampleRate as _), channels: asbd.mChannelsPerFrame as _, data_type: sample_format, }; Ok(format) } } pub fn default_input_format(&self) -> Result { self.default_format(kAudioObjectPropertyScopeInput) } pub fn default_output_format(&self) -> Result { self.default_format(kAudioObjectPropertyScopeOutput) } } impl fmt::Debug for Device { fn fmt(&self, f: &mut fmt::Formatter) -> fmt::Result { f.debug_struct("Device") .field("audio_device_id", &self.audio_device_id) .field("name", &self.name()) .finish() } } // The ID of a stream is its index within the `streams` array of the events loop. #[derive(Debug, Clone, PartialEq, Eq, Hash)] pub struct StreamId(usize); pub struct EventLoop { // This `Arc` is shared with all the callbacks of coreaudio. active_callbacks: Arc, streams: Mutex>>, } struct ActiveCallbacks { // Whenever the `run()` method is called with a callback, this callback is put in this list. callbacks: Mutex>, } struct StreamInner { playing: bool, audio_unit: AudioUnit, // Track the device with which the audio unit was spawned. // // We must do this so that we can avoid changing the device sample rate if there is already // a stream associated with the device. device_id: AudioDeviceID, } // TODO need stronger error identification impl From for BuildStreamError { fn from(err: coreaudio::Error) -> BuildStreamError { match err { coreaudio::Error::RenderCallbackBufferFormatDoesNotMatchAudioUnitStreamFormat | coreaudio::Error::NoKnownSubtype | coreaudio::Error::AudioUnit(coreaudio::error::AudioUnitError::FormatNotSupported) | coreaudio::Error::AudioCodec(_) | coreaudio::Error::AudioFormat(_) => BuildStreamError::FormatNotSupported, _ => BuildStreamError::DeviceNotAvailable, } } } // Create a coreaudio AudioStreamBasicDescription from a CPAL Format. fn asbd_from_format(format: &Format) -> AudioStreamBasicDescription { let n_channels = format.channels as usize; let sample_rate = format.sample_rate.0; let bytes_per_channel = format.data_type.sample_size(); let bits_per_channel = bytes_per_channel * 8; let bytes_per_frame = n_channels * bytes_per_channel; let frames_per_packet = 1; let bytes_per_packet = frames_per_packet * bytes_per_frame; let sample_format = format.data_type; let format_flags = match sample_format { SampleFormat::F32 => (kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked) as u32, _ => kAudioFormatFlagIsPacked as u32, }; let asbd = AudioStreamBasicDescription { mBitsPerChannel: bits_per_channel as _, mBytesPerFrame: bytes_per_frame as _, mChannelsPerFrame: n_channels as _, mBytesPerPacket: bytes_per_packet as _, mFramesPerPacket: frames_per_packet as _, mFormatFlags: format_flags, mFormatID: kAudioFormatLinearPCM, mSampleRate: sample_rate as _, ..Default::default() }; asbd } fn audio_unit_from_device(device: &Device, input: bool) -> Result { let mut audio_unit = { let au_type = if cfg!(target_os = "ios") { // The HalOutput unit isn't available in iOS unfortunately. // RemoteIO is a sensible replacement. // See https://goo.gl/CWwRTx coreaudio::audio_unit::IOType::RemoteIO } else { coreaudio::audio_unit::IOType::HalOutput }; AudioUnit::new(au_type)? }; if input { // Enable input processing. let enable_input = 1u32; audio_unit.set_property( kAudioOutputUnitProperty_EnableIO, Scope::Input, Element::Input, Some(&enable_input), )?; // Disable output processing. let disable_output = 0u32; audio_unit.set_property( kAudioOutputUnitProperty_EnableIO, Scope::Output, Element::Output, Some(&disable_output), )?; } audio_unit.set_property( kAudioOutputUnitProperty_CurrentDevice, Scope::Global, Element::Output, Some(&device.audio_device_id), )?; Ok(audio_unit) } impl EventLoop { #[inline] pub fn new() -> EventLoop { EventLoop { active_callbacks: Arc::new(ActiveCallbacks { callbacks: Mutex::new(Vec::new()) }), streams: Mutex::new(Vec::new()), } } #[inline] pub fn run(&self, mut callback: F) -> ! where F: FnMut(StreamId, StreamData) + Send { { let callback: &mut (FnMut(StreamId, StreamData) + Send) = &mut callback; self.active_callbacks .callbacks .lock() .unwrap() .push(unsafe { mem::transmute(callback) }); } loop { // So the loop does not get optimised out in --release thread::sleep(Duration::new(1u64, 0u32)); } // Note: if we ever change this API so that `run` can return, then it is critical that // we remove the callback from `active_callbacks`. } fn next_stream_id(&self) -> usize { let streams_lock = self.streams.lock().unwrap(); let stream_id = streams_lock .iter() .position(|n| n.is_none()) .unwrap_or(streams_lock.len()); stream_id } // Add the stream to the list of streams within `self`. fn add_stream(&self, stream_id: usize, au: AudioUnit, device_id: AudioDeviceID) { let inner = StreamInner { playing: true, audio_unit: au, device_id: device_id, }; let mut streams_lock = self.streams.lock().unwrap(); if stream_id == streams_lock.len() { streams_lock.push(Some(inner)); } else { streams_lock[stream_id] = Some(inner); } } #[inline] pub fn build_input_stream( &self, device: &Device, format: &Format, ) -> Result { // The scope and element for working with a device's input stream. let scope = Scope::Output; let element = Element::Input; // Check whether or not we need to change the device sample rate to suit the one specified for the stream. unsafe { // Get the current sample rate. let mut property_address = AudioObjectPropertyAddress { mSelector: kAudioDevicePropertyNominalSampleRate, mScope: kAudioObjectPropertyScopeGlobal, mElement: kAudioObjectPropertyElementMaster, }; let sample_rate: f64 = 0.0; let data_size = mem::size_of::() as u32; let status = AudioObjectGetPropertyData( device.audio_device_id, &property_address as *const _, 0, null(), &data_size as *const _ as *mut _, &sample_rate as *const _ as *mut _, ); coreaudio::Error::from_os_status(status)?; // If the requested sample rate is different to the device sample rate, update the device. if sample_rate as u32 != format.sample_rate.0 { // In order to avoid breaking existing input streams we `panic!` if there is already an // active input stream for this device with the actual sample rate. for stream in &*self.streams.lock().unwrap() { if let Some(stream) = stream.as_ref() { if stream.device_id == device.audio_device_id { panic!("cannot change device sample rate for stream as an existing stream \ is already running at the current sample rate."); } } } // 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::(); let mut ranges: Vec = 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::(); 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) { panic!("timeout waiting for sample rate update for device"); } ::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; 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 { 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; 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 }) } } }