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extern crate libc; extern crate aaudio_sys; use std::ffi::c_void; use std::fmt; use std::mem::MaybeUninit; use aaudio_sys as ffi; use ffi::{AAudioStream as AAudioStreamRaw, AAudioStreamBuilder as AAudioStreamBuilderRaw}; #[derive(Debug, Copy, Clone, Eq, PartialEq)] /// These values are returned from AAudio functions to indicate failure. pub enum Error { /// AAudio returned error code that is not a part of this enum Unknown(i32), /// The audio device was disconnected. This could occur, for example, when headphones /// are plugged in or unplugged. The stream cannot be used after the device is disconnected. /// Applications should stop and close the stream. /// If this error is received in an error callback then another thread should be /// used to stop and close the stream. Disconnected, /// An invalid parameter was passed to AAudio. IllegalArgument, /// The requested operation is not appropriate for the current state of AAudio. InvalidState, /// The server rejected the handle used to identify the stream. InvalidHandle, /// The function is not implemented for this stream. Unimplemented, /// A resource or information is unavailable. /// This could occur when an application tries to open too many streams, /// or a timestamp is not available. Unavailable, /// Memory could not be allocated. NoFreeHandles, /// Memory could not be allocated. NoMemory, /// A NULL pointer was passed to AAudio. /// Or a NULL pointer was detected internally. Null, /// An operation took longer than expected. Timeout, WouldBlock, /// The requested data format is not supported. InvalidFormat, /// A requested was out of range. OutOfRange, /// The audio service was not available. NoService, /// The requested sample rate was not supported. InvalidRate, } impl fmt::Display for Error { fn fmt(&self, f: &mut fmt::Formatter) -> fmt::Result { match self { Self::Unknown(code) => write!(f, "Error code {}", code), Self::Disconnected => f.write_str("The audio device was disconnected"), Self::IllegalArgument => f.write_str("An invalid parameter was passed to AAudio."), Self::InvalidState => f.write_str( "The requested operation is not appropriate for the current state of AAudio.", ), Self::InvalidHandle => { f.write_str("The server rejected the handle used to identify the stream.") } Self::Unimplemented => f.write_str("The function is not implemented for this stream."), Self::Unavailable => f.write_str("A resource or information is unavailable."), Self::NoFreeHandles => f.write_str("Memory could not be allocated."), Self::NoMemory => f.write_str("Memory could not be allocated"), Self::Null => f.write_str("A NULL pointer was passed to AAudio."), Self::Timeout => f.write_str("An operation took longer than expected."), Self::WouldBlock => { f.write_str("A blocking operation was invoked where no blocking was expected.") } Self::InvalidFormat => f.write_str("The requested data format is not supported."), Self::OutOfRange => f.write_str("A requested was out of range."), Self::NoService => f.write_str("The audio service was not available."), Self::InvalidRate => f.write_str("The requested sample rate was not supported."), } } } impl Error { fn from_code(code: i32) -> Self { match code { -899 => Self::Disconnected, -898 => Self::IllegalArgument, -895 => Self::InvalidState, -892 => Self::InvalidHandle, -890 => Self::Unimplemented, -889 => Self::Unavailable, -888 => Self::NoFreeHandles, -887 => Self::NoMemory, -886 => Self::Null, -885 => Self::Timeout, -884 => Self::WouldBlock, -883 => Self::InvalidFormat, -882 => Self::OutOfRange, -881 => Self::NoService, -880 => Self::InvalidRate, code => Self::Unknown(code), } } } #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum Direction { /// Audio data will travel out of the device, for example through a speaker. Output, /// Audio data will travel into the device, for example from a microphone. Input, } impl Direction { fn from_i32(val: i32) -> Self { match val { 0 => Self::Output, 1 => Self::Input, direction => panic!("Unexpected direction: {}", direction), } } } /// A sample format. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum Format { Unspecified = 0, /// This format uses the i16 data type. /// The maximum range of the data is -32768 to 32767. I16, /// This format uses the float data type. /// The nominal range of the data is [-1.0f32, 1.0f32). /// Values outside that range may be clipped. /// /// See also 'floatData' at /// https://developer.android.com/reference/android/media/AudioTrack#write(float[],%20int,%20int,%20int) F32, } impl Format { fn from_i32(val: i32) -> Self { match val { 0 => Self::Unspecified, 1 => Self::I16, 2 => Self::F32, format => panic!("Unexpected format: {}", format), } } fn sample_size(&self) -> i32 { match self { Self::Unspecified => 0, Self::I16 => 2, Self::F32 => 4, } } } #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum SharingMode { /// This will be the only stream using a particular source or sink. /// This mode will provide the lowest possible latency. /// You should close Exclusive streams immediately when you are not using them. Exclusive, /// Multiple applications will be mixed by the AAudio Server. /// This will have higher latency than the Exclusive mode. Shared, } impl SharingMode { fn from_i32(val: i32) -> Self { match val { 0 => Self::Exclusive, 1 => Self::Shared, mode => panic!("Unexpected sharing mode: {}", mode), } } } /// The Usage attribute expresses "why" you are playing a sound, what is this sound used for. /// This information is used by certain platforms or routing policies /// to make more refined volume or routing decisions. /// /// Note that these match the equivalent values in android.media.AudioAttributes /// in the Android Java API. /// /// Added in API level 28. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum Usage { /// Use this for streaming media, music performance, video, podcasts, etcetera. Media = 1, /// Use this for voice over IP, telephony, etcetera. VoiceCommunication = 2, /// Use this for sounds associated with telephony such as busy tones, DTMF, etcetera. VoiceCommunicationSignalling = 3, /// Use this to demand the users attention. Alarm = 4, /// Use this for notifying the user when a message has arrived or some /// other background event has occured. Notification = 5, /// Use this when the phone rings. NotificationRingtone = 6, /// Use this to attract the users attention when, for example, the battery is low. NotificationEvent = 10, /// Use this for screen readers, etcetera. AssistanceAccessibility = 11, /// Use this for driving or navigation directions. AssistanceNavigationGuidance = 12, /// Use this for user interface sounds, beeps, etcetera. AssistanceSonification = 13, /// Use this for game audio and sound effects. Game = 14, /// Use this for audio responses to user queries, audio instructions or help utterances. Assistant = 16, /// Use this in case of playing sounds in an emergency. /// Privileged MODIFY_AUDIO_ROUTING permission required. Emergency = 1000, /// Use this for safety sounds and alerts, for example backup camera obstacle detection. /// Privileged MODIFY_AUDIO_ROUTING permission required. Safety = 1001, /// Use this for vehicle status alerts and information, for example the check engine light. /// Privileged MODIFY_AUDIO_ROUTING permission required. VehicleStatus = 1002, /// Use this for traffic announcements, etc. /// Privileged MODIFY_AUDIO_ROUTING permission required. Announcement = 1003, } /// Defines the audio source. /// An audio source defines both a default physical source of audio signal, and a recording /// configuration. /// /// Note that these match the equivalent values in MediaRecorder.AudioSource in the Android Java API. /// /// Added in API level 28. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum InputPreset { /// Use this preset when other presets do not apply. Generic = 1, /// Use this preset when recording video. Camcorder = 5, /// Use this preset when doing speech recognition. VoiceRecognition = 6, /// Use this preset when doing telephony or voice messaging. VoiceCommunication = 7, /// Use this preset to obtain an input with no effects. /// Note that this input will not have automatic gain control /// so the recorded volume may be very low. Unprocessed = 9, /// Use this preset for capturing audio meant to be processed in real time /// and played back for live performance (e.g karaoke). /// The capture path will minimize latency and coupling with playback path. /// Available since API level 29. VoicePerformance = 10, } /// The ContentType attribute describes "what" you are playing. /// It expresses the general category of the content. This information is optional. /// But in case it is known (for instance `Movie` for a /// movie streaming service or `Speech` for /// an audio book application) this information might be used by the audio framework to /// enforce audio focus. /// /// Note that these match the equivalent values in android.media.AudioAttributes /// in the Android Java API. /// /// Added in API level 28. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum ContentType { /// Use this for spoken voice, audio books, etcetera. Speech = 1, /// Use this for pre-recorded or live music. Music = 2, /// Use this for a movie or video soundtrack. Movie = 3, /// Use this for sound is designed to accompany a user action, /// such as a click or beep sound made when the user presses a button. Sonification = 4, } /// Specifying if audio may or may not be captured by other apps or the system. /// /// Note that these match the equivalent values in android.media.AudioAttributes /// in the Android Java API. /// /// Added in API level 29. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum AllowedCapturePolicy { /// Indicates that the audio may be captured by any app. /// /// For privacy, the following usages can not be recorded: `VoiceCommunication*`, /// `Notification*`, `Assistance*` and `Assistant`. /// /// On Android Q, this means only `Media` and `Game` may be captured. /// /// See android.media.AudioAttributes#ALLOW_CAPTURE_BY_ALL. AllowCaptureByAll = 1, /// Indicates that the audio may only be captured by system apps. /// /// System apps can capture for many purposes like accessibility, user guidance... /// but have strong restriction. See /// android.media.AudioAttributes#ALLOW_CAPTURE_BY_SYSTEM for what the system apps /// can do with the capture audio. AllowCaptureBySystem = 2, /// Indicates that the audio may not be recorded by any app, even if it is a system app. /// /// It is encouraged to use `AllowCaptureBySystem` instead of this value as system apps /// provide significant and useful features for the user (eg. accessibility). /// See android.media.AudioAttributes#ALLOW_CAPTURE_BY_NONE. AllowCaptureByNone = 3, } #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum PerformanceMode { /// No particular performance needs. Default. None = 10, /// Extending battery life is more important than low latency. /// /// This mode is not supported in input streams. /// For input, mode NONE will be used if this is requested. PowerSaving = 11, /// Reducing latency is more important than battery life. LowLatency = 12, } impl PerformanceMode { fn from_i32(val: i32) -> Self { match val { 11 => Self::PowerSaving, 12 => Self::LowLatency, _ => Self::None, } } } /// Value returned the data callback function. #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum CallbackResult { /// Continue calling the callback. Continue, /// Stop calling the callback. /// /// The application will still need to call `AAudioStream_requestPause()` /// or `AAudioStream_requestStop()`. Stop, } fn wrap_result(result: i32) -> Result<(), Error> { if result < 0 { Err(Error::from_code(result)) } else { Ok(()) } } struct StreamCallbacks { _data_callback: Box<Box<dyn FnMut(&AAudioStreamInfo, &mut [u8], i32) -> CallbackResult + Send + 'static>>, _error_callback: Box<Box<dyn FnMut(&AAudioStreamInfo, Error) + Send + 'static>>, } #[derive(Debug, Copy, Clone, Eq, PartialEq)] pub enum StreamState { Uninitialized, Unknown, Open, Starting, Started, Pausing, Paused, Flushing, Flushed, Stopping, Stopped, Closing, Closed, Disconnected, } impl StreamState { fn from_i32(val: i32) -> Self { match val { 0 => Self::Uninitialized, 1 => Self::Unknown, 2 => Self::Open, 3 => Self::Starting, 4 => Self::Started, 5 => Self::Pausing, 6 => Self::Paused, 7 => Self::Flushing, 8 => Self::Flushed, 9 => Self::Stopping, 10 => Self::Stopped, 11 => Self::Closing, 12 => Self::Closed, 13 => Self::Disconnected, state => panic!("Unexpected stream state: {}", state), } } } pub struct Timestamp { pub frame_position: i64, pub time_nanos: i64, } pub struct AAudioStream { raw: *mut AAudioStreamRaw, _callbacks: Option<StreamCallbacks>, } unsafe impl Send for AAudioStream {} fn get_timestamp_monotonic(raw: *mut AAudioStreamRaw) -> Result<Timestamp, Error> { let mut frame_position = MaybeUninit::uninit(); let mut time_nanos = MaybeUninit::uninit(); let result = unsafe { ffi::AAudioStream_getTimestamp( raw, libc::CLOCK_MONOTONIC, frame_position.as_mut_ptr(), time_nanos.as_mut_ptr(), ) }; wrap_result(result)?; Ok(unsafe { Timestamp { frame_position: frame_position.assume_init(), time_nanos: time_nanos.assume_init(), } }) } impl AAudioStream { /// Returns the actual sample rate. /// /// Available since API level 26. pub fn get_sample_rate(&self) -> i32 { unsafe { ffi::AAudioStream_getSampleRate(self.raw) } } /// A stream has one or more channels of data. /// A frame will contain one sample for each channel. /// /// Available since API level 26. pub fn get_channel_count(&self) -> i32 { unsafe { ffi::AAudioStream_getChannelCount(self.raw) } } /// Query the maximum number of frames that can be filled without blocking. /// /// Available since API level 26. pub fn get_buffer_size_in_frames(&self) -> i32 { unsafe { ffi::AAudioStream_getBufferSizeInFrames(self.raw) } } /// Query the number of frames that the application should read or write at /// one time for optimal performance. It is OK if an application writes /// a different number of frames. But the buffer size may need to be larger /// in order to avoid underruns or overruns. /// /// Note that this may or may not match the actual device burst size. /// For some endpoints, the burst size can vary dynamically. /// But these tend to be devices with high latency. /// /// Available since API level 26. pub fn get_frames_per_burst(&self) -> i32 { unsafe { ffi::AAudioStream_getFramesPerBurst(self.raw) } } /// Query maximum buffer capacity in frames. /// /// Available since API level 26. pub fn get_buffer_capacity_in_frames(&self) -> i32 { unsafe { ffi::AAudioStream_getBufferCapacityInFrames(self.raw) } } /// Query the size of the buffer that will be passed to the dataProc callback /// in the numFrames parameter. /// /// This call can be used if the application needs to know the value of numFrames before /// the stream is started. This is not normally necessary. /// /// If a specific size was requested by calling /// `AAudioStreamBuilder::set_frames_per_data_callback()` then this will be the same size. /// /// If `AAudioStreamBuilder_set_frames_per_data_callback()` was not called then this will /// return the size chosen by AAudio, or 0. /// /// 0 indicates that the callback buffer size for this stream /// may vary from one dataProc callback to the next. /// /// Available since API level 26. pub fn get_frames_per_data_callback(&self) -> i32 { unsafe { ffi::AAudioStream_getFramesPerDataCallback(self.raw) } } /// An XRun is an Underrun or an Overrun. /// During playing, an underrun will occur if the stream is not written in time /// and the system runs out of valid data. /// During recording, an overrun will occur if the stream is not read in time /// and there is no place to put the incoming data so it is discarded. /// /// An underrun or overrun can cause an audible "pop" or "glitch". /// /// Note that some INPUT devices may not support this function. /// In that case a 0 will always be returned. /// /// Available since API level 26. pub fn get_x_run_count(&self) -> i32 { unsafe { ffi::AAudioStream_getXRunCount(self.raw) } } /// Returns the actual device ID. /// /// Available since API level 26. pub fn get_device_id(&self) -> i32 { unsafe { ffi::AAudioStream_getDeviceId(self.raw) } } /// Returns the actual data format. /// /// Available since API level 26. pub fn get_format(&self) -> Format { let val = unsafe { ffi::AAudioStream_getFormat(self.raw) }; Format::from_i32(val) } /// Provide actual sharing mode. /// /// Available since API level 26. pub fn get_sharing_mode(&self) -> SharingMode { let val = unsafe { ffi::AAudioStream_getSharingMode(self.raw) }; SharingMode::from_i32(val) } /// Get the performance mode used by the stream. /// /// Available since API level 26. pub fn get_performance_mode(&self) -> PerformanceMode { let val = unsafe { ffi::AAudioStream_getPerformanceMode(self.raw) }; PerformanceMode::from_i32(val) } /// Available since API level 26. pub fn get_direction(&self) -> Direction { let val = unsafe { ffi::AAudioStream_getDirection(self.raw) }; Direction::from_i32(val) } /// Returns the number of frames that have been written since the stream was created. /// For an output stream, this will be advanced by the application calling write() /// or by a data callback. /// For an input stream, this will be advanced by the endpoint. /// /// The frame position is monotonically increasing. /// /// Available since API level 26. pub fn get_frames_written(&self) -> i64 { unsafe { ffi::AAudioStream_getFramesWritten(self.raw) } } /// Returns the number of frames that have been read since the stream was created. /// For an output stream, this will be advanced by the endpoint. /// For an input stream, this will be advanced by the application calling read() /// or by a data callback. /// /// The frame position is monotonically increasing. /// /// Available since API level 26. pub fn get_frames_read(&self) -> i64 { unsafe { ffi::AAudioStream_getFramesRead(self.raw) } } /// Passes back the session ID associated with this stream. /// /// The session ID can be used to associate a stream with effects processors. /// The effects are controlled using the Android AudioEffect Java API. /// /// If `AAudioStreamBuilder::set_session_id()` was called with 0 /// then a new session ID should be allocated once when the stream is opened. /// /// If `AAudioStreamBuilder::set_session_id()` was called with a previously allocated /// session ID then that value should be returned. /// /// If `AAudioStreamBuilder::set_session_id()` was not called then this function should /// return -1. /// /// The sessionID for a stream should not change once the stream has been opened. /// /// Available since API level 28. pub fn get_session_id(&self) -> i32 { unsafe { ffi::AAudioStream_getSessionId(self.raw) } } /// Returns the time at which a particular frame was presented. /// This can be used to synchronize audio with video or MIDI. /// It can also be used to align a recorded stream with a playback stream. /// /// Timestamps are only valid when the stream is in `Started` state. /// `InvalidState` will be returned if the stream is not started. /// Note that because request_start() is asynchronous, timestamps will not be valid until /// a short time after calling request_start(). /// So `InvalidState` should not be considered a fatal error. /// Just try calling again later. /// /// If an error occurs, then the position and time will not be modified. /// /// The position and time passed back are monotonically increasing. /// /// Available since API level 26. pub fn get_timestamp_monotonic(&self) -> Result<Timestamp, Error> { get_timestamp_monotonic(self.raw) } /// Query the current state of the client, eg. `Pausing`. /// /// This function will immediately return the state without updating the state. /// If you want to update the client state based on the server state then /// call `AAudioStream::wait_for_state_change()` with currentState /// set to `Unknown` and a zero timeout. /// /// Available since API level 26. pub fn get_state(&self) -> StreamState { let val = unsafe { ffi::AAudioStream_getState(self.raw) }; StreamState::from_i32(val) } /// Free the audio resources associated with the stream. /// /// After this call, the stream will be in `Closing` state. /// /// This function is useful if you want to release the audio resources immediately, /// but still allow queries to the stream to occur from other threads. This often /// happens if you are monitoring stream progress from a UI thread. /// /// Available since API level 30. pub fn release(&mut self) -> Result<(), Error> { let val = unsafe { ffi::AAudioStream_release(self.raw) }; wrap_result(val) } /// Asynchronously request to start playing the stream. For output streams, one should /// write to the stream to fill the buffer before starting. /// Otherwise it will underflow. /// After this call the state will be in `Starting` or `Started`. /// /// Returns 0 for OK or a negative error. /// /// Available since API level 26. pub fn request_start(&mut self) -> Result<(), Error> { let val = unsafe { ffi::AAudioStream_requestStart(self.raw) }; wrap_result(val) } /// Asynchronous request for the stream to pause. /// Pausing a stream will freeze the data flow but not flush any buffers. /// Use `AAudioStream::request_start()` to resume playback after a pause. /// After this call the state will be in `Pausing` or /// `Paused`. /// /// This will return `Unimplemented` for input streams. /// For input streams use `AAudioStream::request_stop()`. /// /// Available since API level 26. pub fn request_pause(&mut self) -> Result<(), Error> { let val = unsafe { ffi::AAudioStream_requestPause(self.raw) }; wrap_result(val) } /// Asynchronous request for the stream to flush. /// Flushing will discard any pending data. /// This call only works if the stream is pausing or paused. /// Frame counters are not reset by a flush. They may be advanced. /// After this call the state will be in `Flushing` or `Flushed`. /// /// This will return `Unimplemented` for input streams. /// /// Available since API level 26. pub fn request_flush(&mut self) -> Result<(), Error> { let val = unsafe { ffi::AAudioStream_requestFlush(self.raw) }; wrap_result(val) } /// Asynchronous request for the stream to stop. /// The stream will stop after all of the data currently buffered has been played. /// After this call the state will be in `Stopping` or `Stopped`. /// /// Available since API level 26. pub fn request_stop(&mut self) -> Result<(), Error> { let val = unsafe { ffi::AAudioStream_requestStop(self.raw) }; wrap_result(val) } /// Wait until the current state no longer matches the input state. /// /// This will update the current client state. /// /// Returns the new state. /// /// Available since API level 26. pub fn wait_for_state_change( &mut self, input_state: StreamState, timeout_nanos: i64, ) -> Result<StreamState, Error> { let mut new_state = MaybeUninit::uninit(); let result = unsafe { ffi::AAudioStream_waitForStateChange( self.raw, input_state as i32, new_state.as_mut_ptr(), timeout_nanos, ) }; wrap_result(result)?; Ok(unsafe { StreamState::from_i32(new_state.assume_init()) }) } /// Read data from the stream. /// Returns the number of frames actually read or a negative error. /// /// The call will wait until the read is complete or until it runs out of time. /// If timeoutNanos is zero then this call will not wait. /// /// Note that timeoutNanoseconds is a relative duration in wall clock time. /// Time will not stop if the thread is asleep. /// So it will be implemented using CLOCK_BOOTTIME. /// /// This call is "strong non-blocking" unless it has to wait for data. /// /// If the call times out then zero or a partial frame count will be returned. /// /// Available since API level 26. /// /// # Arguments /// /// * `buffer` - The slice with the samples. /// * `num_frames` - Number of frames to read. Only complete frames will be written. /// * `timeout_nanoseconds` - Maximum number of nanoseconds to wait for completion. pub fn read( &mut self, buffer: &mut [u8], num_frames: i32, timeout_nanoseconds: i64, ) -> Result<u32, Error> { let result = unsafe { ffi::AAudioStream_read( self.raw, buffer.as_mut_ptr() as *mut c_void, num_frames, timeout_nanoseconds, ) }; wrap_result(result)?; Ok(result as u32) } /// Write data to the stream. /// Returns the number of frames actually written or a negative error. /// /// The call will wait until the write is complete or until it runs out of time. /// If timeoutNanos is zero then this call will not wait. /// /// Note that timeoutNanoseconds is a relative duration in wall clock time. /// Time will not stop if the thread is asleep. /// So it will be implemented using CLOCK_BOOTTIME. /// /// This call is "strong non-blocking" unless it has to wait for room in the buffer. /// /// If the call times out then zero or a partial frame count will be returned. /// /// Available since API level 26. /// /// # Arguments /// /// * `buffer` - The address of the first sample. /// * `num_frames` - Number of frames to write. Only complete frames will be written. /// * `timeout_nanoseconds` - Maximum number of nanoseconds to wait for completion. pub fn write( &mut self, buffer: &[u8], num_frames: i32, timeout_nanoseconds: i64, ) -> Result<u32, Error> { let result = unsafe { ffi::AAudioStream_write( self.raw, buffer.as_ptr() as *const c_void, num_frames, timeout_nanoseconds, ) }; wrap_result(result)?; Ok(result as u32) } /// This can be used to adjust the latency of the buffer by changing /// the threshold where blocking will occur. /// By combining this with `AAudioStream::get_x_run_count()`, the latency can be tuned /// at run-time for each device. /// Returns actual buffer size in frames or a negative error. /// /// This cannot be set higher than `AAudioStream::get_buffer_capacity_in_frames()`. /// /// Note that you will probably not get the exact size you request. /// You can check the return value or call `AAudioStream::get_buffer_size_in_frames()` /// to see what the actual final size is. /// /// Available since API level 26. /// /// # Arguments /// /// * `num_frames` - requested number of frames that can be filled without blocking pub fn set_buffer_size_in_frames(&mut self, num_frames: i32) -> Result<(), Error> { let result = unsafe { ffi::AAudioStream_setBufferSizeInFrames(self.raw, num_frames) }; wrap_result(result) } } impl Drop for AAudioStream { fn drop(&mut self) { unsafe { ffi::AAudioStream_close(self.raw); } } } /// Passed as a callback parameter, providing operations that are safe /// to perform from callback invocation. pub struct AAudioStreamInfo { raw: *mut AAudioStreamRaw, } unsafe impl Send for AAudioStreamInfo {} impl AAudioStreamInfo { /// Returns the actual sample rate. /// /// Available since API level 26. pub fn get_sample_rate(&self) -> i32 { unsafe { ffi::AAudioStream_getSampleRate(self.raw) } } /// A stream has one or more channels of data. /// A frame will contain one sample for each channel. /// /// Available since API level 26. pub fn get_channel_count(&self) -> i32 { unsafe { ffi::AAudioStream_getChannelCount(self.raw) } } /// Query the maximum number of frames that can be filled without blocking. /// /// Available since API level 26. pub fn get_buffer_size_in_frames(&self) -> i32 { unsafe { ffi::AAudioStream_getBufferSizeInFrames(self.raw) } } /// Query the number of frames that the application should read or write at /// one time for optimal performance. It is OK if an application writes /// a different number of frames. But the buffer size may need to be larger /// in order to avoid underruns or overruns. /// /// Note that this may or may not match the actual device burst size. /// For some endpoints, the burst size can vary dynamically. /// But these tend to be devices with high latency. /// /// Available since API level 26. pub fn get_frames_per_burst(&self) -> i32 { unsafe { ffi::AAudioStream_getFramesPerBurst(self.raw) } } /// Query maximum buffer capacity in frames. /// /// Available since API level 26. pub fn get_buffer_capacity_in_frames(&self) -> i32 { unsafe { ffi::AAudioStream_getBufferCapacityInFrames(self.raw) } } /// Query the size of the buffer that will be passed to the dataProc callback /// in the numFrames parameter. /// /// This call can be used if the application needs to know the value of numFrames before /// the stream is started. This is not normally necessary. /// /// If a specific size was requested by calling /// `AAudioStreamBuilder::set_frames_per_data_callback()` then this will be the same size. /// /// If `AAudioStreamBuilder::set_frames_per_data_callback()` was not called then this will /// return the size chosen by AAudio, or 0. /// /// 0 indicates that the callback buffer size for this stream /// may vary from one dataProc callback to the next. /// /// Available since API level 26. pub fn get_frames_per_data_callback(&self) -> i32 { unsafe { ffi::AAudioStream_getFramesPerDataCallback(self.raw) } } /// An XRun is an Underrun or an Overrun. /// During playing, an underrun will occur if the stream is not written in time /// and the system runs out of valid data. /// During recording, an overrun will occur if the stream is not read in time /// and there is no place to put the incoming data so it is discarded. /// /// An underrun or overrun can cause an audible "pop" or "glitch". /// /// Note that some INPUT devices may not support this function. /// In that case a 0 will always be returned. /// /// Available since API level 26. pub fn get_x_run_count(&self) -> i32 { unsafe { ffi::AAudioStream_getXRunCount(self.raw) } } /// Returns the actual device ID. /// /// Available since API level 26. pub fn get_device_id(&self) -> i32 { unsafe { ffi::AAudioStream_getDeviceId(self.raw) } } /// Returns the actual data format. /// /// Available since API level 26. pub fn get_format(&self) -> Format { let val = unsafe { ffi::AAudioStream_getFormat(self.raw) }; Format::from_i32(val) } /// Provide actual sharing mode. /// /// Available since API level 26. pub fn get_sharing_mode(&self) -> SharingMode { let val = unsafe { ffi::AAudioStream_getSharingMode(self.raw) }; SharingMode::from_i32(val) } /// Get the performance mode used by the stream. /// /// Available since API level 26. pub fn get_performance_mode(&self) -> PerformanceMode { let val = unsafe { ffi::AAudioStream_getPerformanceMode(self.raw) }; PerformanceMode::from_i32(val) } /// Available since API level 26. pub fn get_direction(&self) -> Direction { let val = unsafe { ffi::AAudioStream_getDirection(self.raw) }; Direction::from_i32(val) } /// Returns the number of frames that have been written since the stream was created. /// For an output stream, this will be advanced by the application calling `write()` /// or by a data callback. /// For an input stream, this will be advanced by the endpoint. /// /// The frame position is monotonically increasing. /// /// Available since API level 26. pub fn get_frames_written(&self) -> i64 { unsafe { ffi::AAudioStream_getFramesWritten(self.raw) } } /// Returns the number of frames that have been read since the stream was created. /// For an output stream, this will be advanced by the endpoint. /// For an input stream, this will be advanced by the application calling `read()` /// or by a data callback. /// /// The frame position is monotonically increasing. /// /// Available since API level 26. pub fn get_frames_read(&self) -> i64 { unsafe { ffi::AAudioStream_getFramesRead(self.raw) } } /// Passes back the session ID associated with this stream. /// /// The session ID can be used to associate a stream with effects processors. /// The effects are controlled using the Android AudioEffect Java API. /// /// If `AAudioStreamBuilder::set_session_id()` was called with 0 /// then a new session ID should be allocated once when the stream is opened. /// /// If `AAudioStreamBuilder::set_session_id()` was called with a previously allocated /// session ID then that value should be returned. /// /// If `AAudioStreamBuilder::set_session_id()` was not called then this function should /// return -1. /// /// The sessionID for a stream should not change once the stream has been opened. /// /// Available since API level 28. pub fn get_session_id(&self) -> i32 { unsafe { ffi::AAudioStream_getSessionId(self.raw) } } /// Returns the time at which a particular frame was presented. /// This can be used to synchronize audio with video or MIDI. /// It can also be used to align a recorded stream with a playback stream. /// /// Timestamps are only valid when the stream is in `Started` state. /// `InvalidState` will be returned if the stream is not started. /// Note that because request_start() is asynchronous, timestamps will not be valid until /// a short time after calling request_start(). /// So `InvalidState` should not be considered a fatal error. /// Just try calling again later. /// /// If an error occurs, then the position and time will not be modified. /// /// The position and time passed back are monotonically increasing. /// /// Available since API level 26. pub fn get_timestamp_monotonic(&self) -> Result<Timestamp, Error> { get_timestamp_monotonic(self.raw) } /// Query the current state of the client, eg. `Pausing`. /// /// This function will immediately return the state without updating the state. /// If you want to update the client state based on the server state then /// call AAudioStream::wait_for_state_change() with currentState /// set to `Unknown` and a zero timeout. /// /// Available since API level 26. pub fn get_state(&self) -> StreamState { let val = unsafe { ffi::AAudioStream_getState(self.raw) }; StreamState::from_i32(val) } } pub struct AAudioStreamBuilder { raw: *mut AAudioStreamBuilderRaw, callbacks: Option<StreamCallbacks>, } unsafe extern "C" fn raw_data_callback( stream: *mut AAudioStreamRaw, user_data: *mut c_void, audio_data: *mut c_void, num_frames: i32, ) -> i32 { match std::panic::catch_unwind(|| { let stream = AAudioStreamInfo { raw: stream }; let data: &mut [u8] = std::slice::from_raw_parts_mut( audio_data as *mut u8, (num_frames * stream.get_channel_count() * stream.get_format().sample_size()) as usize, ); let callback = user_data as *mut Box<dyn FnMut(&AAudioStreamInfo, &mut [u8], i32) -> CallbackResult>; let callback = &mut *callback; callback(&stream, data, num_frames) as i32 }) { Ok(r) => r, Err(e) => { eprintln!("{:?}", e); std::process::abort(); } } } unsafe extern "C" fn raw_error_callback( stream: *mut AAudioStreamRaw, user_data: *mut c_void, error: i32, ) { if let Err(e) = std::panic::catch_unwind(|| { let stream = AAudioStreamInfo { raw: stream }; let callback = user_data as *mut Box<dyn FnMut(&AAudioStreamInfo, Error)>; let callback = &mut *callback; callback(&stream, Error::from_code(error)); }) { eprintln!("{:?}", e); std::process::abort(); } } impl AAudioStreamBuilder { pub fn new() -> Result<Self, Error> { let mut raw = MaybeUninit::<*mut AAudioStreamBuilderRaw>::uninit(); let result = unsafe { ffi::AAudio_createStreamBuilder(raw.as_mut_ptr()) }; wrap_result(result)?; Ok(Self { raw: unsafe { raw.assume_init() }, callbacks: None, }) } /// Request that AAudio call the `data_callback` when the stream is running and the /// `error_callback` if any error occurs or the stream is disconnected. /// /// Note that when using data callback, the audio data will be passed in or out /// of the function as an argument. /// So you cannot call `AAudioStream::write()` or `AAudioStream::read()` /// on the same stream that has an active data callback. /// /// The data callback function will start being called after `AAudioStream::request_start()` /// is called. /// It will stop being called after `AAudioStream::request_pause()` or /// `AAudioStream::request_stop()` is called. /// /// The `data_callback` function will be called on a real-time thread owned by AAudio. /// Note that numFrames can vary unless `AAudioStreamBuilder::set_frames_per_data_callback()` /// is called. /// /// Also note that this callback function should be considered a "real-time" function. /// It must not do anything that could cause an unbounded delay because that can cause the /// audio to glitch or pop. /// /// These are things the function should NOT do: /// * allocate memory using, for example, malloc() or new /// * any file operations such as opening, closing, reading or writing /// * any network operations such as streaming /// * use any mutexes or other synchronization primitives /// * sleep /// * stop or close the stream /// * `AAudioStream::read()` /// * `AAudioStream::write()` /// /// If you need to move data, eg. MIDI commands, in or out of the callback function then /// we recommend the use of non-blocking techniques such as an atomic FIFO. /// /// The `error_callback` will be called, for example, if a headset or a USB device is unplugged causing the stream's /// device to be unavailable or "disconnected". /// Another possible cause of error would be a timeout or an unanticipated internal error. /// /// In response, this function should signal or create another thread to stop /// and close this stream. The other thread could then reopen a stream on another device. /// Do not stop or close the stream, or reopen the new stream, directly from this callback. /// /// The `error_callback` will not be called because of actions by the application, such as stopping /// or closing a stream. /// /// Note that the AAudio callbacks will never be called simultaneously from multiple threads. /// /// Available since API level 26. pub fn set_callbacks<D, E>(mut self, data_callback: D, error_callback: E) -> Self where D: FnMut(&AAudioStreamInfo, &mut [u8], i32) -> CallbackResult + Send + 'static, E: FnMut(&AAudioStreamInfo, Error) + Send + 'static, { let data_callback: Box< Box<dyn FnMut(&AAudioStreamInfo, &mut [u8], i32) -> CallbackResult + Send + 'static>, > = Box::new(Box::new(data_callback)); let error_callback: Box<Box<dyn FnMut(&AAudioStreamInfo, Error) + Send + 'static>> = Box::new(Box::new(error_callback)); let data_callback_raw = Box::into_raw(data_callback); let error_callback_raw = Box::into_raw(error_callback); let callbacks = StreamCallbacks { _data_callback: unsafe { Box::from_raw(data_callback_raw) }, _error_callback: unsafe { Box::from_raw(error_callback_raw) }, }; unsafe { ffi::AAudioStreamBuilder_setDataCallback( self.raw, Some(raw_data_callback), data_callback_raw as *mut c_void, ); ffi::AAudioStreamBuilder_setErrorCallback( self.raw, Some(raw_error_callback), error_callback_raw as *mut c_void, ); } self.callbacks = Some(callbacks); self } /// Request an audio device identified device using an ID. /// On Android, for example, the ID could be obtained from the Java AudioManager. /// /// The default, if you do not call this function, is 0, /// in which case the primary device will be used. /// /// Available since API level 26. /// /// # Arguments /// /// * `device_id` - device identifier or 0 for unspecified pub fn set_device_id(self, device_id: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setDeviceId(self.raw, device_id); } self } /// Request a sample rate in Hertz. /// /// The default, if you do not call this function, is 0 (unspecified). /// An optimal value will then be chosen when the stream is opened. /// After opening a stream with an unspecified value, the application must /// query for the actual value, which may vary by device. /// /// If an exact value is specified then an opened stream will use that value. /// If a stream cannot be opened with the specified value then the open will fail. /// /// Available since API level 26. /// /// # Arguments /// /// * `sample_rate` - frames per second. Common rates include 44100 and 48000 Hz. pub fn set_sample_rate(self, sample_rate: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setSampleRate(self.raw, sample_rate); } self } /// Request a number of channels for the stream. /// /// The default, if you do not call this function, is unspecified. /// An optimal value will then be chosen when the stream is opened. /// After opening a stream with an unspecified value, the application must /// query for the actual value, which may vary by device. /// /// If an exact value is specified then an opened stream will use that value. /// If a stream cannot be opened with the specified value then the open will fail. /// /// Available since API level 26. /// /// # Arguments /// /// * `channel_count` - Number of channels desired. pub fn set_channel_count(self, channel_count: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setChannelCount(self.raw, channel_count); } self } /// Request a sample data format, for example `Format::I16`. /// /// The default, if you do not call this function, is `Unspecified`. /// An optimal value will then be chosen when the stream is opened. /// After opening a stream with an unspecified value, the application must /// query for the actual value, which may vary by device. /// /// If an exact value is specified then an opened stream will use that value. /// If a stream cannot be opened with the specified value then the open will fail. /// /// Available since API level 26. /// /// # Arguments /// /// * `format` - the sample data format. pub fn set_format(self, format: Format) -> Self { unsafe { ffi::AAudioStreamBuilder_setFormat(self.raw, format as i32) } self } /// Request a mode for sharing the device. /// /// The default, if you do not call this function, is `SharingMode::Shared`. /// /// The requested sharing mode may not be available. /// The application can query for the actual mode after the stream is opened. /// /// Available since API level 26. /// /// # Arguments /// /// * `sharing_mode` - `SharingMode::Shared` or `SharingMode::Exclusive` pub fn set_sharing_mode(self, sharing_mode: SharingMode) -> Self { unsafe { ffi::AAudioStreamBuilder_setSharingMode(self.raw, sharing_mode as i32) } self } /// Request the direction for a stream. /// /// The default, if you do not call this function, is `Direction::Output`. /// /// Available since API level 26. /// /// # Arguments /// /// * `direction` - `Direction::Output` or `Direction::Input` pub fn set_direction(self, direction: Direction) -> Self { unsafe { ffi::AAudioStreamBuilder_setDirection(self.raw, direction as i32) } self } /// Set the requested buffer capacity in frames. /// The final AAudioStream capacity may differ, but will probably be at least this big. /// /// The default, if you do not call this function, is unspecified. /// /// Available since API level 26. /// /// # Arguments /// /// * `num_frames` - the desired buffer capacity in frames or 0 for unspecified pub fn set_buffer_capacity_in_frames(self, num_frames: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setBufferCapacityInFrames(self.raw, num_frames as i32) } self } /// Set the requested performance mode. /// /// Supported modes are None, PowerSaving and LowLatency. /// /// The default, if you do not call this function, is None. /// /// You may not get the mode you requested. /// You can call `AAudioStream::get_performance_mode()` /// to find out the final mode for the stream. /// /// Available since API level 26. /// /// # Arguments /// /// * `mode` - the desired performance mode, eg. LowLatency pub fn set_performance_mode(self, mode: PerformanceMode) -> Self { unsafe { ffi::AAudioStreamBuilder_setPerformanceMode(self.raw, mode as i32) } self } /// Set the intended use case for the stream. /// /// The AAudio system will use this information to optimize the /// behavior of the stream. /// This could, for example, affect how volume and focus is handled for the stream. /// /// The default, if you do not call this function, is `Usage::Media`. /// /// Available since API level 28. /// /// * `usage` - the desired usage, eg. `Usage::Game` pub fn set_usage(self, usage: Usage) -> Self { unsafe { ffi::AAudioStreamBuilder_setUsage(self.raw, usage as i32) } self } /// Set the type of audio data that the stream will carry. /// /// The AAudio system will use this information to optimize the /// behavior of the stream. /// This could, for example, affect whether a stream is paused when a notification occurs. /// /// The default, if you do not call this function, is `ContentType::Music`. /// /// Available since API level 28. /// /// # Arguments /// /// * `content_type` - the type of audio data, eg. `ContentType::Speech` pub fn set_content_type(self, content_type: ContentType) -> Self { unsafe { ffi::AAudioStreamBuilder_setContentType(self.raw, content_type as i32) } self } /// Set the input (capture) preset for the stream. /// /// The AAudio system will use this information to optimize the /// behavior of the stream. /// This could, for example, affect which microphones are used and how the /// recorded data is processed. /// /// The default, if you do not call this function, is `InputPreset::VoiceRecognition`. /// That is because `InputPreset::VoiceRecognition` is the preset with the lowest latency /// on many platforms. /// /// Available since API level 28. /// /// # Arguments /// /// * `input_preset` - the desired configuration for recording pub fn set_input_preset(self, input_preset: InputPreset) -> Self { unsafe { ffi::AAudioStreamBuilder_setInputPreset(self.raw, input_preset as i32) } self } /// Specify whether this stream audio may or may not be captured by other apps or the system. /// /// The default is `AllowedCapturePolicy::AllowCaptureByAll`. /// /// Note that an application can also set its global policy, in which case the most restrictive /// policy is always applied. See android.media.AudioAttributes#setAllowedCapturePolicy(int) /// /// Available since API level 29. /// /// # Arguments /// /// * `policy` - the desired level of opt-out from being captured. pub fn set_allowed_capture_policy(self, policy: AllowedCapturePolicy) -> Self { unsafe { ffi::AAudioStreamBuilder_setAllowedCapturePolicy(self.raw, policy as i32) } self } /// Equivalent to invoking `AAudioStreamBuilder::set_session_id` with 0 argument. pub fn allocate_session_id(self) -> Self { unsafe { ffi::AAudioStreamBuilder_setSessionId(self.raw, 0) } self } /// Equivalent to invoking `AAudioStreamBuilder::set_session_id` with -1 argument. pub fn remove_session_id(self) -> Self { unsafe { ffi::AAudioStreamBuilder_setSessionId(self.raw, -1) } self } /// The session ID can be used to associate a stream with effects processors. /// The effects are controlled using the Android AudioEffect Java API. /// /// The default, if you do not call this function, is -1 (none). /// /// If set to 0 then a session ID will be allocated /// when the stream is opened. /// /// The allocated session ID can be obtained by calling `AAudioStream::get_session_id()` /// and then used with this function when opening another stream. /// This allows effects to be shared between streams. /// /// Session IDs from AAudio can be used with the Android Java APIs and vice versa. /// So a session ID from an AAudio stream can be passed to Java /// and effects applied using the Java AudioEffect API. /// /// Note that allocating or setting a session ID may result in a stream with higher latency. /// /// Allocated session IDs will always be positive and nonzero. /// /// Available since API level 28. /// /// # Arguments /// /// * `session_id` - an allocated sessionID or 0 to allocate a new sessionID pub fn set_session_id(self, session_id: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setSessionId(self.raw, session_id as i32) } self } /// Indicates whether this input stream must be marked as privacy sensitive or not. /// /// When true, this input stream is privacy sensitive and any concurrent capture /// is not permitted. /// /// This is off (false) by default except when the input preset is `InputPreset::VoiceCommunication` /// or `InputPreset::Camcorder`. /// /// Always takes precedence over default from input preset when set explicitly. /// /// Only relevant if the stream direction is `Direction::Input`. /// /// Added in API level 30. /// /// # Arguments /// /// * `privacy_sensitive` - `true` if capture from this stream must be marked as privacy sensitive, `false` otherwise. pub fn set_privacy_sensitive(self, privacy_sensitive: bool) -> Self { unsafe { ffi::AAudioStreamBuilder_setPrivacySensitive(self.raw, privacy_sensitive) } self } /// Set the requested data callback buffer size in frames. /// See [`set_callbacks`]. /// /// The default, if you do not call this function, is unspecified. /// /// For the lowest possible latency, do not call this function. AAudio will then /// call the dataProc callback function with whatever size is optimal. /// That size may vary from one callback to another. /// /// Only use this function if the application requires a specific number of frames for processing. /// The application might, for example, be using an FFT that requires /// a specific power-of-two sized buffer. /// /// AAudio may need to add additional buffering in order to adapt between the internal /// buffer size and the requested buffer size. /// /// If you do call this function then the requested size should be less than /// half the buffer capacity, to allow double buffering. /// /// Available since API level 26. /// /// * `num_frames` - the desired buffer size in frames or 0 for unspecified /// /// [`set_callbacks`]: AAudioStreamBuilder::set_callbacks pub fn set_frames_per_data_callback(self, num_frames: i32) -> Self { unsafe { ffi::AAudioStreamBuilder_setFramesPerDataCallback(self.raw, num_frames) } self } /// Open a stream based on the options in the AAudioStreamBuilder. pub fn open_stream(mut self) -> Result<AAudioStream, Error> { let mut raw = MaybeUninit::<*mut AAudioStreamRaw>::uninit(); let result = unsafe { ffi::AAudioStreamBuilder_openStream(self.raw, raw.as_mut_ptr()) }; wrap_result(result)?; let stream = AAudioStream { raw: unsafe { raw.assume_init() }, _callbacks: self.callbacks.take(), }; Ok(stream) } } impl Drop for AAudioStreamBuilder { fn drop(&mut self) { unsafe { ffi::AAudioStreamBuilder_delete(self.raw); } } }