/*
* Copyright (C) 2016 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
/**
* @addtogroup Audio
* @{
*/
/**
* @file AAudio.h
*/
/**
* This is the 'C' API for AAudio.
*/
#ifndef AAUDIO_AAUDIO_H
#define AAUDIO_AAUDIO_H
#include <time.h>
#ifdef __cplusplus
extern "C" {
#endif
/**
* This is used to represent a value that has not been specified.
* For example, an application could use {@link #AAUDIO_UNSPECIFIED} to indicate
* that is did not not care what the specific value of a parameter was
* and would accept whatever it was given.
*/
#define AAUDIO_UNSPECIFIED 0
enum {
/**
* Audio data will travel out of the device, for example through a speaker.
*/
AAUDIO_DIRECTION_OUTPUT,
/**
* Audio data will travel into the device, for example from a microphone.
*/
AAUDIO_DIRECTION_INPUT
};
typedef int32_t aaudio_direction_t;
enum {
AAUDIO_FORMAT_INVALID = -1,
AAUDIO_FORMAT_UNSPECIFIED = 0,
/**
* This format uses the int16_t data type.
* The maximum range of the data is -32768 to 32767.
*/
AAUDIO_FORMAT_PCM_I16,
/**
* This format uses the float data type.
* The nominal range of the data is [-1.0f, 1.0f).
* Values outside that range may be clipped.
*
* See also 'floatData' at
* https://developer.android.com/reference/android/media/AudioTrack#write(float[],%20int,%20int,%20int)
*/
AAUDIO_FORMAT_PCM_FLOAT
};
typedef int32_t aaudio_format_t;
/**
* These result codes are returned from AAudio functions to indicate success or failure.
* Note that error return codes may change in the future so applications should generally
* not rely on specific return codes.
*/
enum {
/**
* The call was successful.
*/
AAUDIO_OK,
AAUDIO_ERROR_BASE = -900, // TODO review
/**
* 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.
*/
AAUDIO_ERROR_DISCONNECTED,
/**
* An invalid parameter was passed to AAudio.
*/
AAUDIO_ERROR_ILLEGAL_ARGUMENT,
// reserved
AAUDIO_ERROR_INTERNAL = AAUDIO_ERROR_ILLEGAL_ARGUMENT + 2,
/**
* The requested operation is not appropriate for the current state of AAudio.
*/
AAUDIO_ERROR_INVALID_STATE,
// reserved
// reserved
/* The server rejected the handle used to identify the stream.
*/
AAUDIO_ERROR_INVALID_HANDLE = AAUDIO_ERROR_INVALID_STATE + 3,
// reserved
/**
* The function is not implemented for this stream.
*/
AAUDIO_ERROR_UNIMPLEMENTED = AAUDIO_ERROR_INVALID_HANDLE + 2,
/**
* A resource or information is unavailable.
* This could occur when an application tries to open too many streams,
* or a timestamp is not available.
*/
AAUDIO_ERROR_UNAVAILABLE,
AAUDIO_ERROR_NO_FREE_HANDLES,
/**
* Memory could not be allocated.
*/
AAUDIO_ERROR_NO_MEMORY,
/**
* A NULL pointer was passed to AAudio.
* Or a NULL pointer was detected internally.
*/
AAUDIO_ERROR_NULL,
/**
* An operation took longer than expected.
*/
AAUDIO_ERROR_TIMEOUT,
AAUDIO_ERROR_WOULD_BLOCK,
/**
* The requested data format is not supported.
*/
AAUDIO_ERROR_INVALID_FORMAT,
/**
* A requested was out of range.
*/
AAUDIO_ERROR_OUT_OF_RANGE,
/**
* The audio service was not available.
*/
AAUDIO_ERROR_NO_SERVICE,
/**
* The requested sample rate was not supported.
*/
AAUDIO_ERROR_INVALID_RATE
};
typedef int32_t aaudio_result_t;
enum
{
AAUDIO_STREAM_STATE_UNINITIALIZED = 0,
AAUDIO_STREAM_STATE_UNKNOWN,
AAUDIO_STREAM_STATE_OPEN,
AAUDIO_STREAM_STATE_STARTING,
AAUDIO_STREAM_STATE_STARTED,
AAUDIO_STREAM_STATE_PAUSING,
AAUDIO_STREAM_STATE_PAUSED,
AAUDIO_STREAM_STATE_FLUSHING,
AAUDIO_STREAM_STATE_FLUSHED,
AAUDIO_STREAM_STATE_STOPPING,
AAUDIO_STREAM_STATE_STOPPED,
AAUDIO_STREAM_STATE_CLOSING,
AAUDIO_STREAM_STATE_CLOSED,
AAUDIO_STREAM_STATE_DISCONNECTED
};
typedef int32_t aaudio_stream_state_t;
enum {
/**
* 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.
*/
AAUDIO_SHARING_MODE_EXCLUSIVE,
/**
* Multiple applications will be mixed by the AAudio Server.
* This will have higher latency than the EXCLUSIVE mode.
*/
AAUDIO_SHARING_MODE_SHARED
};
typedef int32_t aaudio_sharing_mode_t;
enum {
/**
* No particular performance needs. Default.
*/
AAUDIO_PERFORMANCE_MODE_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.
*/
AAUDIO_PERFORMANCE_MODE_POWER_SAVING,
/**
* Reducing latency is more important than battery life.
*/
AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
};
typedef int32_t aaudio_performance_mode_t;
/**
* 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 {@link android.media.AudioAttributes}
* in the Android Java API.
*
* Added in API level 28.
*/
enum {
/**
* Use this for streaming media, music performance, video, podcasts, etcetera.
*/
AAUDIO_USAGE_MEDIA = 1,
/**
* Use this for voice over IP, telephony, etcetera.
*/
AAUDIO_USAGE_VOICE_COMMUNICATION = 2,
/**
* Use this for sounds associated with telephony such as busy tones, DTMF, etcetera.
*/
AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3,
/**
* Use this to demand the users attention.
*/
AAUDIO_USAGE_ALARM = 4,
/**
* Use this for notifying the user when a message has arrived or some
* other background event has occured.
*/
AAUDIO_USAGE_NOTIFICATION = 5,
/**
* Use this when the phone rings.
*/
AAUDIO_USAGE_NOTIFICATION_RINGTONE = 6,
/**
* Use this to attract the users attention when, for example, the battery is low.
*/
AAUDIO_USAGE_NOTIFICATION_EVENT = 10,
/**
* Use this for screen readers, etcetera.
*/
AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11,
/**
* Use this for driving or navigation directions.
*/
AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12,
/**
* Use this for user interface sounds, beeps, etcetera.
*/
AAUDIO_USAGE_ASSISTANCE_SONIFICATION = 13,
/**
* Use this for game audio and sound effects.
*/
AAUDIO_USAGE_GAME = 14,
/**
* Use this for audio responses to user queries, audio instructions or help utterances.
*/
AAUDIO_USAGE_ASSISTANT = 16
};
typedef int32_t aaudio_usage_t;
/**
* The CONTENT_TYPE 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 AAUDIO_CONTENT_TYPE_MOVIE for a
* movie streaming service or AAUDIO_CONTENT_TYPE_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 {@link android.media.AudioAttributes}
* in the Android Java API.
*
* Added in API level 28.
*/
enum {
/**
* Use this for spoken voice, audio books, etcetera.
*/
AAUDIO_CONTENT_TYPE_SPEECH = 1,
/**
* Use this for pre-recorded or live music.
*/
AAUDIO_CONTENT_TYPE_MUSIC = 2,
/**
* Use this for a movie or video soundtrack.
*/
AAUDIO_CONTENT_TYPE_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.
*/
AAUDIO_CONTENT_TYPE_SONIFICATION = 4
};
typedef int32_t aaudio_content_type_t;
/**
* 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.
*/
enum {
/**
* Use this preset when other presets do not apply.
*/
AAUDIO_INPUT_PRESET_GENERIC = 1,
/**
* Use this preset when recording video.
*/
AAUDIO_INPUT_PRESET_CAMCORDER = 5,
/**
* Use this preset when doing speech recognition.
*/
AAUDIO_INPUT_PRESET_VOICE_RECOGNITION = 6,
/**
* Use this preset when doing telephony or voice messaging.
*/
AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION = 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.
*/
AAUDIO_INPUT_PRESET_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.
*/
AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE = 10,
};
typedef int32_t aaudio_input_preset_t;
/**
* Specifying if audio may or may not be captured by other apps or the system.
*
* Note that these match the equivalent values in {@link android.media.AudioAttributes}
* in the Android Java API.
*
* Added in API level 29.
*/
enum {
/**
* Indicates that the audio may be captured by any app.
*
* For privacy, the following usages can not be recorded: AAUDIO_VOICE_COMMUNICATION*,
* AAUDIO_USAGE_NOTIFICATION*, AAUDIO_USAGE_ASSISTANCE* and {@link #AAUDIO_USAGE_ASSISTANT}.
*
* On {@link android.os.Build.VERSION_CODES#Q}, this means only {@link #AAUDIO_USAGE_MEDIA}
* and {@link #AAUDIO_USAGE_GAME} may be captured.
*
* See {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_ALL}.
*/
AAUDIO_ALLOW_CAPTURE_BY_ALL = 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
* {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_SYSTEM} for what the system apps
* can do with the capture audio.
*/
AAUDIO_ALLOW_CAPTURE_BY_SYSTEM = 2,
/**
* Indicates that the audio may not be recorded by any app, even if it is a system app.
*
* It is encouraged to use {@link #AAUDIO_ALLOW_CAPTURE_BY_SYSTEM} instead of this value as system apps
* provide significant and useful features for the user (eg. accessibility).
* See {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_NONE}.
*/
AAUDIO_ALLOW_CAPTURE_BY_NONE = 3,
};
typedef int32_t aaudio_allowed_capture_policy_t;
/**
* These may be used with AAudioStreamBuilder_setSessionId().
*
* Added in API level 28.
*/
enum {
/**
* Do not allocate a session ID.
* Effects cannot be used with this stream.
* Default.
*
* Added in API level 28.
*/
AAUDIO_SESSION_ID_NONE = -1,
/**
* Allocate a session ID that can be used to attach and control
* effects using the Java AudioEffects API.
* Note that using this may result in higher latency.
*
* Note that this matches the value of AudioManager.AUDIO_SESSION_ID_GENERATE.
*
* Added in API level 28.
*/
AAUDIO_SESSION_ID_ALLOCATE = 0,
};
typedef int32_t aaudio_session_id_t;
typedef struct AAudioStreamStruct AAudioStream;
typedef struct AAudioStreamBuilderStruct AAudioStreamBuilder;
#ifndef AAUDIO_API
#define AAUDIO_API /* export this symbol */
#endif
// ============================================================
// Audio System
// ============================================================
/**
* The text is the ASCII symbol corresponding to the returnCode,
* or an English message saying the returnCode is unrecognized.
* This is intended for developers to use when debugging.
* It is not for display to users.
*
* @return pointer to a text representation of an AAudio result code.
*/
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) __INTRODUCED_IN(26);
/**
* The text is the ASCII symbol corresponding to the stream state,
* or an English message saying the state is unrecognized.
* This is intended for developers to use when debugging.
* It is not for display to users.
*
* @return pointer to a text representation of an AAudio state.
*/
AAUDIO_API const char * AAudio_convertStreamStateToText(aaudio_stream_state_t state)
__INTRODUCED_IN(26);
// ============================================================
// StreamBuilder
// ============================================================
/**
* Create a StreamBuilder that can be used to open a Stream.
*
* The deviceId is initially unspecified, meaning that the current default device will be used.
*
* The default direction is {@link #AAUDIO_DIRECTION_OUTPUT}.
* The default sharing mode is {@link #AAUDIO_SHARING_MODE_SHARED}.
* The data format, samplesPerFrames and sampleRate are unspecified and will be
* chosen by the device when it is opened.
*
* AAudioStreamBuilder_delete() must be called when you are done using the builder.
*/
AAUDIO_API aaudio_result_t AAudio_createStreamBuilder(AAudioStreamBuilder** builder)
__INTRODUCED_IN(26);
/**
* 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 {@link #AAUDIO_UNSPECIFIED},
* in which case the primary device will be used.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param deviceId device identifier or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setDeviceId(AAudioStreamBuilder* builder,
int32_t deviceId) __INTRODUCED_IN(26);
/**
* Request a sample rate in Hertz.
*
* The default, if you do not call this function, is {@link #AAUDIO_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.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sampleRate frames per second. Common rates include 44100 and 48000 Hz.
*/
AAUDIO_API void AAudioStreamBuilder_setSampleRate(AAudioStreamBuilder* builder,
int32_t sampleRate) __INTRODUCED_IN(26);
/**
* Request a number of channels for the stream.
*
* The default, if you do not call this function, is {@link #AAUDIO_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.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param channelCount Number of channels desired.
*/
AAUDIO_API void AAudioStreamBuilder_setChannelCount(AAudioStreamBuilder* builder,
int32_t channelCount) __INTRODUCED_IN(26);
/**
* Identical to AAudioStreamBuilder_setChannelCount().
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param samplesPerFrame Number of samples in a frame.
*/
AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
int32_t samplesPerFrame) __INTRODUCED_IN(26);
/**
* Request a sample data format, for example {@link #AAUDIO_FORMAT_PCM_I16}.
*
* The default, if you do not call this function, is {@link #AAUDIO_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.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param format common formats are {@link #AAUDIO_FORMAT_PCM_FLOAT} and
* {@link #AAUDIO_FORMAT_PCM_I16}.
*/
AAUDIO_API void AAudioStreamBuilder_setFormat(AAudioStreamBuilder* builder,
aaudio_format_t format) __INTRODUCED_IN(26);
/**
* Request a mode for sharing the device.
*
* The default, if you do not call this function, is {@link #AAUDIO_SHARING_MODE_SHARED}.
*
* The requested sharing mode may not be available.
* The application can query for the actual mode after the stream is opened.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sharingMode {@link #AAUDIO_SHARING_MODE_SHARED} or {@link #AAUDIO_SHARING_MODE_EXCLUSIVE}
*/
AAUDIO_API void AAudioStreamBuilder_setSharingMode(AAudioStreamBuilder* builder,
aaudio_sharing_mode_t sharingMode) __INTRODUCED_IN(26);
/**
* Request the direction for a stream.
*
* The default, if you do not call this function, is {@link #AAUDIO_DIRECTION_OUTPUT}.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param direction {@link #AAUDIO_DIRECTION_OUTPUT} or {@link #AAUDIO_DIRECTION_INPUT}
*/
AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
aaudio_direction_t direction) __INTRODUCED_IN(26);
/**
* 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 {@link #AAUDIO_UNSPECIFIED}.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer capacity in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder,
int32_t numFrames) __INTRODUCED_IN(26);
/**
* Set the requested performance mode.
*
* Supported modes are {@link #AAUDIO_PERFORMANCE_MODE_NONE},
* {@link #AAUDIO_PERFORMANCE_MODE_POWER_SAVING} * and {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}.
*
* The default, if you do not call this function, is {@link #AAUDIO_PERFORMANCE_MODE_NONE}.
*
* You may not get the mode you requested.
* You can call AAudioStream_getPerformanceMode()
* to find out the final mode for the stream.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param mode the desired performance mode, eg. {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}
*/
AAUDIO_API void AAudioStreamBuilder_setPerformanceMode(AAudioStreamBuilder* builder,
aaudio_performance_mode_t mode) __INTRODUCED_IN(26);
/**
* 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 {@link #AAUDIO_USAGE_MEDIA}.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param usage the desired usage, eg. {@link #AAUDIO_USAGE_GAME}
*/
AAUDIO_API void AAudioStreamBuilder_setUsage(AAudioStreamBuilder* builder,
aaudio_usage_t usage) __INTRODUCED_IN(28);
/**
* 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 {@link #AAUDIO_CONTENT_TYPE_MUSIC}.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param contentType the type of audio data, eg. {@link #AAUDIO_CONTENT_TYPE_SPEECH}
*/
AAUDIO_API void AAudioStreamBuilder_setContentType(AAudioStreamBuilder* builder,
aaudio_content_type_t contentType) __INTRODUCED_IN(28);
/**
* 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 {@link #AAUDIO_INPUT_PRESET_VOICE_RECOGNITION}.
* That is because VOICE_RECOGNITION is the preset with the lowest latency
* on many platforms.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired configuration for recording
*/
AAUDIO_API void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder* builder,
aaudio_input_preset_t inputPreset) __INTRODUCED_IN(28);
/**
* Specify whether this stream audio may or may not be captured by other apps or the system.
*
* The default is {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}.
*
* Note that an application can also set its global policy, in which case the most restrictive
* policy is always applied. See {@link android.media.AudioAttributes#setAllowedCapturePolicy(int)}
*
* Added in API level 29.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired level of opt-out from being captured.
*/
AAUDIO_API void AAudioStreamBuilder_setAllowedCapturePolicy(AAudioStreamBuilder* builder,
aaudio_allowed_capture_policy_t capturePolicy) __INTRODUCED_IN(29);
/** Set the requested session ID.
*
* 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 {@link #AAUDIO_SESSION_ID_NONE}.
*
* If set to {@link #AAUDIO_SESSION_ID_ALLOCATE} then a session ID will be allocated
* when the stream is opened.
*
* The allocated session ID can be obtained by calling AAudioStream_getSessionId()
* 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.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sessionId an allocated sessionID or {@link #AAUDIO_SESSION_ID_ALLOCATE}
*/
AAUDIO_API void AAudioStreamBuilder_setSessionId(AAudioStreamBuilder* builder,
aaudio_session_id_t sessionId) __INTRODUCED_IN(28);
/**
* Return one of these values from the data callback function.
*/
enum {
/**
* Continue calling the callback.
*/
AAUDIO_CALLBACK_RESULT_CONTINUE = 0,
/**
* Stop calling the callback.
*
* The application will still need to call AAudioStream_requestPause()
* or AAudioStream_requestStop().
*/
AAUDIO_CALLBACK_RESULT_STOP,
};
typedef int32_t aaudio_data_callback_result_t;
/**
* Prototype for the data function that is passed to AAudioStreamBuilder_setDataCallback().
*
* For an output stream, this function should render and write numFrames of data
* in the streams current data format to the audioData buffer.
*
* For an input stream, this function should read and process numFrames of data
* from the audioData buffer.
*
* The audio data is passed through the buffer. So do NOT call AAudioStream_read() or
* AAudioStream_write() on the stream that is making the callback.
*
* Note that numFrames can vary unless AAudioStreamBuilder_setFramesPerDataCallback()
* 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:
* <ul>
* <li>allocate memory using, for example, malloc() or new</li>
* <li>any file operations such as opening, closing, reading or writing</li>
* <li>any network operations such as streaming</li>
* <li>use any mutexes or other synchronization primitives</li>
* <li>sleep</li>
* <li>stop or close the stream</li>
* <li>AAudioStream_read()</li>
* <li>AAudioStream_write()</li>
* </ul>
*
* The following are OK to call from the data callback:
* <ul>
* <li>AAudioStream_get*()</li>
* <li>AAudio_convertResultToText()</li>
* </ul>
*
* 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param userData the same address that was passed to AAudioStreamBuilder_setCallback()
* @param audioData a pointer to the audio data
* @param numFrames the number of frames to be processed, which can vary
* @return AAUDIO_CALLBACK_RESULT_*
*/
typedef aaudio_data_callback_result_t (*AAudioStream_dataCallback)(
AAudioStream *stream,
void *userData,
void *audioData,
int32_t numFrames);
/**
* Request that AAudio call this functions when the stream is running.
*
* Note that when using this 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 callback function will start being called after AAudioStream_requestStart()
* is called.
* It will stop being called after AAudioStream_requestPause() or
* AAudioStream_requestStop() is called.
*
* This callback function will be called on a real-time thread owned by AAudio. See
* {@link #AAudioStream_dataCallback} for more information.
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will process audio data.
* @param userData pointer to an application data structure that will be passed
* to the callback functions.
*/
AAUDIO_API void AAudioStreamBuilder_setDataCallback(AAudioStreamBuilder* builder,
AAudioStream_dataCallback callback, void *userData) __INTRODUCED_IN(26);
/**
* Set the requested data callback buffer size in frames.
* See {@link #AAudioStream_dataCallback}.
*
* The default, if you do not call this function, is {@link #AAUDIO_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.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setFramesPerDataCallback(AAudioStreamBuilder* builder,
int32_t numFrames) __INTRODUCED_IN(26);
/**
* Prototype for the callback function that is passed to
* AAudioStreamBuilder_setErrorCallback().
*
* The following may NOT be called from the error callback:
* <ul>
* <li>AAudioStream_requestStop()</li>
* <li>AAudioStream_requestPause()</li>
* <li>AAudioStream_close()</li>
* <li>AAudioStream_waitForStateChange()</li>
* <li>AAudioStream_read()</li>
* <li>AAudioStream_write()</li>
* </ul>
*
* The following are OK to call from the error callback:
* <ul>
* <li>AAudioStream_get*()</li>
* <li>AAudio_convertResultToText()</li>
* </ul>
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param userData the same address that was passed to AAudioStreamBuilder_setErrorCallback()
* @param error an AAUDIO_ERROR_* value.
*/
typedef void (*AAudioStream_errorCallback)(
AAudioStream *stream,
void *userData,
aaudio_result_t error);
/**
* Request that AAudio call this function if any error occurs or the stream is disconnected.
*
* It 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.
*
* This 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.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will be called if an error occurs.
* @param userData pointer to an application data structure that will be passed
* to the callback functions.
*/
AAUDIO_API void AAudioStreamBuilder_setErrorCallback(AAudioStreamBuilder* builder,
AAudioStream_errorCallback callback, void *userData) __INTRODUCED_IN(26);
/**
* Open a stream based on the options in the StreamBuilder.
*
* AAudioStream_close() must be called when finished with the stream to recover
* the memory and to free the associated resources.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param stream pointer to a variable to receive the new stream reference
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStreamBuilder_openStream(AAudioStreamBuilder* builder,
AAudioStream** stream) __INTRODUCED_IN(26);
/**
* Delete the resources associated with the StreamBuilder.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStreamBuilder_delete(AAudioStreamBuilder* builder)
__INTRODUCED_IN(26);
// ============================================================
// Stream Control
// ============================================================
/**
* Free the resources associated with a stream created by AAudioStreamBuilder_openStream()
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_close(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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 {@link #AAUDIO_STREAM_STATE_STARTING} or
* {@link #AAUDIO_STREAM_STATE_STARTED}.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestStart(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Asynchronous request for the stream to pause.
* Pausing a stream will freeze the data flow but not flush any buffers.
* Use AAudioStream_requestStart() to resume playback after a pause.
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_PAUSING} or
* {@link #AAUDIO_STREAM_STATE_PAUSED}.
*
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
* For input streams use AAudioStream_requestStop().
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestPause(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Asynchronous request for the stream to flush.
* Flushing will discard any pending data.
* This call only works if the stream is pausing or paused. TODO review
* Frame counters are not reset by a flush. They may be advanced.
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_FLUSHING} or
* {@link #AAUDIO_STREAM_STATE_FLUSHED}.
*
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestFlush(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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 {@link #AAUDIO_STREAM_STATE_STOPPING} or
* {@link #AAUDIO_STREAM_STATE_STOPPED}.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestStop(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Query the current state of the client, eg. {@link #AAUDIO_STREAM_STATE_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_waitForStateChange() with currentState
* set to {@link #AAUDIO_STREAM_STATE_UNKNOWN} and a zero timeout.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_stream_state_t AAudioStream_getState(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Wait until the current state no longer matches the input state.
*
* This will update the current client state.
*
* <pre><code>
* aaudio_result_t result = AAUDIO_OK;
* aaudio_stream_state_t currentState = AAudioStream_getState(stream);
* aaudio_stream_state_t inputState = currentState;
* while (result == AAUDIO_OK && currentState != AAUDIO_STREAM_STATE_PAUSED) {
* result = AAudioStream_waitForStateChange(
* stream, inputState, ¤tState, MY_TIMEOUT_NANOS);
* inputState = currentState;
* }
* </code></pre>
*
* @param stream A reference provided by AAudioStreamBuilder_openStream()
* @param inputState The state we want to avoid.
* @param nextState Pointer to a variable that will be set to the new state.
* @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion.
* @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_waitForStateChange(AAudioStream* stream,
aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState,
int64_t timeoutNanoseconds) __INTRODUCED_IN(26);
// ============================================================
// Stream I/O
// ============================================================
/**
* Read data from the stream.
*
* 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.
*
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to read. Only complete frames will be written.
* @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion.
* @return The number of frames actually read or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_read(AAudioStream* stream,
void *buffer, int32_t numFrames, int64_t timeoutNanoseconds) __INTRODUCED_IN(26);
/**
* Write data to the stream.
*
* 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.
*
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to write. Only complete frames will be written.
* @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion.
* @return The number of frames actually written or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_write(AAudioStream* stream,
const void *buffer, int32_t numFrames, int64_t timeoutNanoseconds) __INTRODUCED_IN(26);
// ============================================================
// Stream - queries
// ============================================================
/**
* This can be used to adjust the latency of the buffer by changing
* the threshold where blocking will occur.
* By combining this with AAudioStream_getXRunCount(), the latency can be tuned
* at run-time for each device.
*
* This cannot be set higher than AAudioStream_getBufferCapacityInFrames().
*
* Note that you will probably not get the exact size you request.
* You can check the return value or call AAudioStream_getBufferSizeInFrames()
* to see what the actual final size is.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param numFrames requested number of frames that can be filled without blocking
* @return actual buffer size in frames or a negative error
*/
AAUDIO_API aaudio_result_t AAudioStream_setBufferSizeInFrames(AAudioStream* stream,
int32_t numFrames) __INTRODUCED_IN(26);
/**
* Query the maximum number of frames that can be filled without blocking.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer size in frames.
*/
AAUDIO_API int32_t AAudioStream_getBufferSizeInFrames(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return burst size
*/
AAUDIO_API int32_t AAudioStream_getFramesPerBurst(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Query maximum buffer capacity in frames.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer capacity in frames
*/
AAUDIO_API int32_t AAudioStream_getBufferCapacityInFrames(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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_setFramesPerDataCallback() then this will be the same size.
*
* If AAudioStreamBuilder_setFramesPerDataCallback() was not called then this will
* return the size chosen by AAudio, or {@link #AAUDIO_UNSPECIFIED}.
*
* {@link #AAUDIO_UNSPECIFIED} indicates that the callback buffer size for this stream
* may vary from one dataProc callback to the next.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return callback buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API int32_t AAudioStream_getFramesPerDataCallback(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the underrun or overrun count
*/
AAUDIO_API int32_t AAudioStream_getXRunCount(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sample rate
*/
AAUDIO_API int32_t AAudioStream_getSampleRate(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* A stream has one or more channels of data.
* A frame will contain one sample for each channel.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of channels
*/
AAUDIO_API int32_t AAudioStream_getChannelCount(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Identical to AAudioStream_getChannelCount().
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of samples frame
*/
AAUDIO_API int32_t AAudioStream_getSamplesPerFrame(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual device ID
*/
AAUDIO_API int32_t AAudioStream_getDeviceId(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual data format
*/
AAUDIO_API aaudio_format_t AAudioStream_getFormat(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Provide actual sharing mode.
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sharing mode
*/
AAUDIO_API aaudio_sharing_mode_t AAudioStream_getSharingMode(AAudioStream* stream)
__INTRODUCED_IN(26);
/**
* Get the performance mode used by the stream.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_performance_mode_t AAudioStream_getPerformanceMode(AAudioStream* stream)
__INTRODUCED_IN(26);
/**
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return direction
*/
AAUDIO_API aaudio_direction_t AAudioStream_getDirection(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Passes back 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames written
*/
AAUDIO_API int64_t AAudioStream_getFramesWritten(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Passes back 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
*/
AAUDIO_API int64_t AAudioStream_getFramesRead(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* 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_setSessionId() was
* called with {@link #AAUDIO_SESSION_ID_ALLOCATE}
* then a new session ID should be allocated once when the stream is opened.
*
* If AAudioStreamBuilder_setSessionId() was called with a previously allocated
* session ID then that value should be returned.
*
* If AAudioStreamBuilder_setSessionId() was not called then this function should
* return {@link #AAUDIO_SESSION_ID_NONE}.
*
* The sessionID for a stream should not change once the stream has been opened.
*
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return session ID or {@link #AAUDIO_SESSION_ID_NONE}
*/
AAUDIO_API aaudio_session_id_t AAudioStream_getSessionId(AAudioStream* stream) __INTRODUCED_IN(28);
/**
* Passes back 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 {@link #AAUDIO_STREAM_STATE_STARTED}.
* {@link #AAUDIO_ERROR_INVALID_STATE} will be returned if the stream is not started.
* Note that because requestStart() is asynchronous, timestamps will not be valid until
* a short time after calling requestStart().
* So {@link #AAUDIO_ERROR_INVALID_STATE} 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.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param clockid CLOCK_MONOTONIC or CLOCK_BOOTTIME
* @param framePosition pointer to a variable to receive the position
* @param timeNanoseconds pointer to a variable to receive the time
* @return {@link #AAUDIO_OK} or a negative error
*/
AAUDIO_API aaudio_result_t AAudioStream_getTimestamp(AAudioStream* stream,
clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds) __INTRODUCED_IN(26);
/**
* Return the use case for the stream.
*
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
*/
AAUDIO_API aaudio_usage_t AAudioStream_getUsage(AAudioStream* stream) __INTRODUCED_IN(28);
/**
* Return the content type for the stream.
*
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return content type, for example {@link #AAUDIO_CONTENT_TYPE_MUSIC}
*/
AAUDIO_API aaudio_content_type_t AAudioStream_getContentType(AAudioStream* stream)
__INTRODUCED_IN(28);
/**
* Return the input preset for the stream.
*
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return input preset, for example {@link #AAUDIO_INPUT_PRESET_CAMCORDER}
*/
AAUDIO_API aaudio_input_preset_t AAudioStream_getInputPreset(AAudioStream* stream)
__INTRODUCED_IN(28);
/**
* Return the policy that determines whether the audio may or may not be captured
* by other apps or the system.
*
* Added in API level 29.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the allowed capture policy, for example {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}
*/
AAUDIO_API aaudio_allowed_capture_policy_t AAudioStream_getAllowedCapturePolicy(
AAudioStream* stream) __INTRODUCED_IN(29);
#ifdef __cplusplus
}
#endif
#endif //AAUDIO_AAUDIO_H
/** @} */