Added in API level 3

AudioTrack

Kotlin |Java

public class AudioTrack
extends Object implements AudioRouting, VolumeAutomation

java.lang.Object
↳	android.media.AudioTrack

The AudioTrack class manages and plays a single audio resource for Java applications. It allows streaming of PCM audio buffers to the audio sink for playback. This is achieved by "pushing" the data to the AudioTrack object using one of the write(byte[], int, int), write(short[], int, int), and write(float[], int, int, int) methods.

An AudioTrack instance can operate under two modes: static or streaming.
In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using one of the write() methods. These are blocking and return when the data has been transferred from the Java layer to the native layer and queued for playback. The streaming mode is most useful when playing blocks of audio data that for instance are:

too big to fit in memory because of the duration of the sound to play,
too big to fit in memory because of the characteristics of the audio data (high sampling rate, bits per sample ...)
received or generated while previously queued audio is playing.

The static mode should be chosen when dealing with short sounds that fit in memory and that need to be played with the smallest latency possible. The static mode will therefore be preferred for UI and game sounds that are played often, and with the smallest overhead possible.

Upon creation, an AudioTrack object initializes its associated audio buffer. The size of this buffer, specified during the construction, determines how long an AudioTrack can play before running out of data.
For an AudioTrack using the static mode, this size is the maximum size of the sound that can be played from it.
For the streaming mode, data will be written to the audio sink in chunks of sizes less than or equal to the total buffer size. AudioTrack is not final and thus permits subclasses, but such use is not recommended.

Summary

Nested classes
`class`	`AudioTrack.Builder` Builder class for `AudioTrack` objects.
`class`	`AudioTrack.MetricsConstants`
`interface`	`AudioTrack.OnCodecFormatChangedListener` Interface definition for a listener for codec format changes.
`interface`	`AudioTrack.OnPlaybackPositionUpdateListener` Interface definition for a callback to be invoked when the playback head position of an AudioTrack has reached a notification marker or has increased by a certain period.
`interface`	`AudioTrack.OnRoutingChangedListener` This interface was deprecated in API level 24. users should switch to the general purpose `AudioRouting.OnRoutingChangedListener` class instead.
`class`	`AudioTrack.StreamEventCallback` Abstract class to receive event notifications about the stream playback in offloaded mode.

Constants
`int`	`DUAL_MONO_MODE_LL` This mode indicates that a stereo stream should be presented with the left audio channel replicated into the right audio channel.
`int`	`DUAL_MONO_MODE_LR` This mode indicates that a stereo stream should be presented with the left and right audio channels blended together and delivered to both channels.
`int`	`DUAL_MONO_MODE_OFF` This mode disables any Dual Mono presentation effect.
`int`	`DUAL_MONO_MODE_RR` This mode indicates that a stereo stream should be presented with the right audio channel replicated into the left audio channel.
`int`	`ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR` Encapsulation metadata type for DVB AD descriptor.
`int`	`ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER` Encapsulation metadata type for framework tuner information.
`int`	`ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT` Encapsulation metadata type for placement of supplementary audio.
`int`	`ENCAPSULATION_MODE_ELEMENTARY_STREAM` This mode indicates metadata encapsulation with an elementary stream payload.
`int`	`ENCAPSULATION_MODE_NONE` This mode indicates no metadata encapsulation, which is the default mode for sending audio data through `AudioTrack`.
`int`	`ERROR` Denotes a generic operation failure.
`int`	`ERROR_BAD_VALUE` Denotes a failure due to the use of an invalid value.
`int`	`ERROR_DEAD_OBJECT` An error code indicating that the object reporting it is no longer valid and needs to be recreated.
`int`	`ERROR_INVALID_OPERATION` Denotes a failure due to the improper use of a method.
`int`	`MODE_STATIC` Creation mode where audio data is transferred from Java to the native layer only once before the audio starts playing.
`int`	`MODE_STREAM` Creation mode where audio data is streamed from Java to the native layer as the audio is playing.
`int`	`PERFORMANCE_MODE_LOW_LATENCY` Low latency performance mode for an `AudioTrack`.
`int`	`PERFORMANCE_MODE_NONE` Default performance mode for an `AudioTrack`.
`int`	`PERFORMANCE_MODE_POWER_SAVING` Power saving performance mode for an `AudioTrack`.
`int`	`PLAYSTATE_PAUSED` indicates AudioTrack state is paused
`int`	`PLAYSTATE_PLAYING` indicates AudioTrack state is playing
`int`	`PLAYSTATE_STOPPED` indicates AudioTrack state is stopped
`int`	`STATE_INITIALIZED` State of an AudioTrack that is ready to be used.
`int`	`STATE_NO_STATIC_DATA` State of a successfully initialized AudioTrack that uses static data, but that hasn't received that data yet.
`int`	`STATE_UNINITIALIZED` State of an AudioTrack that was not successfully initialized upon creation.
`int`	`SUCCESS` Denotes a successful operation.
`int`	`SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT` Supplementary audio placement left.
`int`	`SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL` Supplementary audio placement normal.
`int`	`SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT` Supplementary audio placement right.
`int`	`WRITE_BLOCKING` The write mode indicating the write operation will block until all data has been written, to be used as the actual value of the writeMode parameter in `write(byte[], int, int, int)`, `write(short[], int, int, int)`, `write(float[], int, int, int)`, `write(java.nio.ByteBuffer, int, int)`, and `write(java.nio.ByteBuffer, int, int, long)`.
`int`	`WRITE_NON_BLOCKING` The write mode indicating the write operation will return immediately after queuing as much audio data for playback as possible without blocking, to be used as the actual value of the writeMode parameter in `write(java.nio.ByteBuffer, int, int)`, `write(short[], int, int, int)`, `write(float[], int, int, int)`, `write(java.nio.ByteBuffer, int, int)`, and `write(java.nio.ByteBuffer, int, int, long)`.

Public constructors
`AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, int bufferSizeInBytes, int mode)` This constructor is deprecated. use `Builder` or `AudioTrack(android.media.AudioAttributes, android.media.AudioFormat, int, int, int)` to specify the `AudioAttributes` instead of the stream type which is only for volume control.
`AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, int bufferSizeInBytes, int mode, int sessionId)` This constructor is deprecated. use `Builder` or `AudioTrack(android.media.AudioAttributes, android.media.AudioFormat, int, int, int)` to specify the `AudioAttributes` instead of the stream type which is only for volume control.
`AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes, int mode, int sessionId)` Class constructor with `AudioAttributes` and `AudioFormat`.

Public methods
`void`	`addOnCodecFormatChangedListener(Executor executor, AudioTrack.OnCodecFormatChangedListener listener)` Adds an `OnCodecFormatChangedListener` to receive notifications of codec format change events on this `AudioTrack`.
`void`	`addOnRoutingChangedListener(AudioTrack.OnRoutingChangedListener listener, Handler handler)` This method was deprecated in API level 24. users should switch to the general purpose `AudioRouting.OnRoutingChangedListener` class instead.
`void`	`addOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener, Handler handler)` Adds an `AudioRouting.OnRoutingChangedListener` to receive notifications of routing changes on this AudioTrack.
`int`	`attachAuxEffect(int effectId)` Attaches an auxiliary effect to the audio track.
`VolumeShaper`	`createVolumeShaper(VolumeShaper.Configuration configuration)` Returns a `VolumeShaper` object that can be used modify the volume envelope of the player or track.
`void`	`flush()` Flushes the audio data currently queued for playback.
`AudioAttributes`	`getAudioAttributes()` Returns the `AudioAttributes` used in configuration.
`float`	`getAudioDescriptionMixLeveldB()` Returns the Audio Description mix level in dB.
`int`	`getAudioFormat()` Returns the configured audio data encoding.
`int`	`getAudioSessionId()` Returns the audio session ID.
`int`	`getBufferCapacityInFrames()` Returns the maximum size of the `AudioTrack` buffer in frames.
`int`	`getBufferSizeInFrames()` Returns the effective size of the `AudioTrack` buffer that the application writes to.
`int`	`getChannelConfiguration()` Returns the configured channel position mask.
`int`	`getChannelCount()` Returns the configured number of channels.
`int`	`getDualMonoMode()` Returns the Dual Mono mode presentation setting.
`AudioFormat`	`getFormat()` Returns the configured `AudioTrack` format.
`LogSessionId`	`getLogSessionId()` Returns the `LogSessionId`.
`static float`	`getMaxVolume()` Returns the maximum gain value, which is greater than or equal to 1.0.
`PersistableBundle`	`getMetrics()` Return Metrics data about the current AudioTrack instance.
`static int`	`getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat)` Returns the estimated minimum buffer size required for an AudioTrack object to be created in the `MODE_STREAM` mode.
`static float`	`getMinVolume()` Returns the minimum gain value, which is the constant 0.0.
`static int`	`getNativeOutputSampleRate(int streamType)` Returns the output sample rate in Hz for the specified stream type.
`int`	`getNotificationMarkerPosition()` Returns marker position expressed in frames.
`int`	`getOffloadDelay()` Return the decoder delay of an offloaded track, expressed in frames, previously set with `setOffloadDelayPadding(int, int)`, or 0 if it was never modified.
`int`	`getOffloadPadding()` Return the decoder padding of an offloaded track, expressed in frames, previously set with `setOffloadDelayPadding(int, int)`, or 0 if it was never modified.
`int`	`getPerformanceMode()` Returns the current performance mode of the `AudioTrack`.
`int`	`getPlayState()` Returns the playback state of the AudioTrack instance.
`int`	`getPlaybackHeadPosition()` Returns the playback head position expressed in frames.
`PlaybackParams`	`getPlaybackParams()` Returns the current playback parameters.
`int`	`getPlaybackRate()` Returns the current playback sample rate rate in Hz.
`int`	`getPositionNotificationPeriod()` Returns the notification update period expressed in frames.
`AudioDeviceInfo`	`getPreferredDevice()` Returns the selected output specified by `setPreferredDevice(AudioDeviceInfo)`.
`AudioDeviceInfo`	`getRoutedDevice()` Returns an `AudioDeviceInfo` identifying the current routing of this AudioTrack.
`int`	`getSampleRate()` Returns the configured audio source sample rate in Hz.
`int`	`getStartThresholdInFrames()` Returns the streaming start threshold of the `AudioTrack`.
`int`	`getState()` Returns the state of the AudioTrack instance.
`int`	`getStreamType()` Returns the volume stream type of this AudioTrack.
`boolean`	`getTimestamp(AudioTimestamp timestamp)` Poll for a timestamp on demand.
`int`	`getUnderrunCount()` Returns the number of underrun occurrences in the application-level write buffer since the AudioTrack was created.
`static boolean`	`isDirectPlaybackSupported(AudioFormat format, AudioAttributes attributes)` This method was deprecated in API level 33. Use `AudioManager#getDirectPlaybackSupport(AudioFormat, AudioAttributes)` instead.
`boolean`	`isOffloadedPlayback()` Returns whether the track was built with `Builder#setOffloadedPlayback(boolean)` set to `true`.
`void`	`pause()` Pauses the playback of the audio data.
`void`	`play()` Starts playing an AudioTrack.
`void`	`registerStreamEventCallback(Executor executor, AudioTrack.StreamEventCallback eventCallback)` Registers a callback for the notification of stream events.
`void`	`release()` Releases the native AudioTrack resources.
`int`	`reloadStaticData()` Sets the playback head position within the static buffer to zero, that is it rewinds to start of static buffer.
`void`	`removeOnCodecFormatChangedListener(AudioTrack.OnCodecFormatChangedListener listener)` Removes an `OnCodecFormatChangedListener` which has been previously added to receive codec format change events.
`void`	`removeOnRoutingChangedListener(AudioTrack.OnRoutingChangedListener listener)` This method was deprecated in API level 24. users should switch to the general purpose `AudioRouting.OnRoutingChangedListener` class instead.
`void`	`removeOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener)` Removes an `AudioRouting.OnRoutingChangedListener` which has been previously added to receive rerouting notifications.
`boolean`	`setAudioDescriptionMixLeveldB(float level)` Sets the Audio Description mix level in dB.
`int`	`setAuxEffectSendLevel(float level)` Sets the send level of the audio track to the attached auxiliary effect `attachAuxEffect(int)`.
`int`	`setBufferSizeInFrames(int bufferSizeInFrames)` Limits the effective size of the `AudioTrack` buffer that the application writes to.
`boolean`	`setDualMonoMode(int dualMonoMode)` Sets the Dual Mono mode presentation on the output device.
`void`	`setLogSessionId(LogSessionId logSessionId)` Sets a `LogSessionId` instance to this AudioTrack for metrics collection.
`int`	`setLoopPoints(int startInFrames, int endInFrames, int loopCount)` Sets the loop points and the loop count.
`int`	`setNotificationMarkerPosition(int markerInFrames)` Sets the position of the notification marker.
`void`	`setOffloadDelayPadding(int delayInFrames, int paddingInFrames)` Configures the delay and padding values for the current compressed stream playing in offload mode.
`void`	`setOffloadEndOfStream()` Declares that the last write() operation on this track provided the last buffer of this stream.
`int`	`setPlaybackHeadPosition(int positionInFrames)` Sets the playback head position within the static buffer.
`void`	`setPlaybackParams(PlaybackParams params)` Sets the playback parameters.
`void`	`setPlaybackPositionUpdateListener(AudioTrack.OnPlaybackPositionUpdateListener listener, Handler handler)` Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update.
`void`	`setPlaybackPositionUpdateListener(AudioTrack.OnPlaybackPositionUpdateListener listener)` Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update.
`int`	`setPlaybackRate(int sampleRateInHz)` Sets the playback sample rate for this track.
`int`	`setPositionNotificationPeriod(int periodInFrames)` Sets the period for the periodic notification event.
`boolean`	`setPreferredDevice(AudioDeviceInfo deviceInfo)` Specifies an audio device (via an `AudioDeviceInfo` object) to route the output from this AudioTrack.
`int`	`setPresentation(AudioPresentation presentation)` Sets the audio presentation.
`int`	`setStartThresholdInFrames(int startThresholdInFrames)` Sets the streaming start threshold for an `AudioTrack`.
`int`	`setStereoVolume(float leftGain, float rightGain)` This method was deprecated in API level 21. Applications should use `setVolume(float)` instead, as it more gracefully scales down to mono, and up to multi-channel content beyond stereo.
`int`	`setVolume(float gain)` Sets the specified output gain value on all channels of this track.
`void`	`stop()` Stops playing the audio data.
`void`	`unregisterStreamEventCallback(AudioTrack.StreamEventCallback eventCallback)` Unregisters the callback for notification of stream events, previously registered with `registerStreamEventCallback(java.util.concurrent.Executor, android.media.AudioTrack.StreamEventCallback)`.
`int`	`write(float[] audioData, int offsetInFloats, int sizeInFloats, int writeMode)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).
`int`	`write(short[] audioData, int offsetInShorts, int sizeInShorts)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).
`int`	`write(byte[] audioData, int offsetInBytes, int sizeInBytes)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).
`int`	`write(ByteBuffer audioData, int sizeInBytes, int writeMode)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).
`int`	`write(ByteBuffer audioData, int sizeInBytes, int writeMode, long timestamp)` Writes the audio data to the audio sink for playback in streaming mode on a HW_AV_SYNC track.
`int`	`write(short[] audioData, int offsetInShorts, int sizeInShorts, int writeMode)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).
`int`	`write(byte[] audioData, int offsetInBytes, int sizeInBytes, int writeMode)` Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode).

Protected methods
`void`	`finalize()` Called by the garbage collector on an object when garbage collection determines that there are no more references to the object.
`int`	`getNativeFrameCount()` This method was deprecated in API level 19. Use the identical public method `getBufferSizeInFrames()` instead.
`void`	`setState(int state)` This method was deprecated in API level 19. Only accessible by subclasses, which are not recommended for AudioTrack.

Inherited methods

From class


        
          java.lang.Object

`Object`	`clone()` Creates and returns a copy of this object.
`boolean`	`equals(Object obj)` Indicates whether some other object is "equal to" this one.
`void`	`finalize()` Called by the garbage collector on an object when garbage collection determines that there are no more references to the object.
`final Class<?>`	`getClass()` Returns the runtime class of this `Object`.
`int`	`hashCode()` Returns a hash code value for the object.
`final void`	`notify()` Wakes up a single thread that is waiting on this object's monitor.
`final void`	`notifyAll()` Wakes up all threads that are waiting on this object's monitor.
`String`	`toString()` Returns a string representation of the object.
`final void`	`wait(long timeoutMillis, int nanos)` Causes the current thread to wait until it is awakened, typically by being notified or interrupted, or until a certain amount of real time has elapsed.
`final void`	`wait(long timeoutMillis)` Causes the current thread to wait until it is awakened, typically by being notified or interrupted, or until a certain amount of real time has elapsed.
`final void`	`wait()` Causes the current thread to wait until it is awakened, typically by being notified or interrupted.

From interface


        
          android.media.AudioRouting

`abstract void`	`addOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener, Handler handler)` Adds an `AudioRouting.OnRoutingChangedListener` to receive notifications of routing changes on this AudioTrack/AudioRecord.
`abstract AudioDeviceInfo`	`getPreferredDevice()` Returns the selected output/input specified by `setPreferredDevice(AudioDeviceInfo)`.
`abstract AudioDeviceInfo`	`getRoutedDevice()` Returns an `AudioDeviceInfo` identifying the current routing of this AudioTrack/AudioRecord.
`abstract void`	`removeOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener)` Removes an `AudioRouting.OnRoutingChangedListener` which has been previously added to receive rerouting notifications.
`abstract boolean`	`setPreferredDevice(AudioDeviceInfo deviceInfo)` Specifies an audio device (via an `AudioDeviceInfo` object) to route the output/input to/from.

From interface


        
          android.media.VolumeAutomation


        abstract
        
        
        
        
        VolumeShaper


      createVolumeShaper(VolumeShaper.Configuration configuration)

Returns a VolumeShaper object that can be used modify the volume envelope of the player or track.

Constants

DUAL_MONO_MODE_LL

Added in API level 30

public static final int DUAL_MONO_MODE_LL

This mode indicates that a stereo stream should be presented with the left audio channel replicated into the right audio channel. Behavior for non-stereo streams is implementation defined. A suggested guideline is that all channels with left-right stereo symmetry will have the left channel position replicated into the right channel position. The center channels (with no left/right symmetry) or unbalanced channels are left alone. The Dual Mono effect occurs before volume scaling.

Constant Value: 2 (0x00000002)

DUAL_MONO_MODE_LR

Added in API level 30

public static final int DUAL_MONO_MODE_LR

This mode indicates that a stereo stream should be presented with the left and right audio channels blended together and delivered to both channels. Behavior for non-stereo streams is implementation defined. A suggested guideline is that the left-right stereo symmetric channels are pairwise blended; the other channels such as center are left alone. The Dual Mono effect occurs before volume scaling.

Constant Value: 1 (0x00000001)

DUAL_MONO_MODE_OFF

Added in API level 30

public static final int DUAL_MONO_MODE_OFF

This mode disables any Dual Mono presentation effect.

Constant Value: 0 (0x00000000)

DUAL_MONO_MODE_RR

Added in API level 30

public static final int DUAL_MONO_MODE_RR

This mode indicates that a stereo stream should be presented with the right audio channel replicated into the left audio channel. Behavior for non-stereo streams is implementation defined. A suggested guideline is that all channels with left-right stereo symmetry will have the right channel position replicated into the left channel position. The center channels (with no left/right symmetry) or unbalanced channels are left alone. The Dual Mono effect occurs before volume scaling.

Constant Value: 3 (0x00000003)

ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR

Added in API level 30

public static final int ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR

Encapsulation metadata type for DVB AD descriptor. This metadata is formatted per ETSI TS 101 154 Table E.1: AD_descriptor.

Constant Value: 2 (0x00000002)

ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER

Added in API level 30

public static final int ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER

Encapsulation metadata type for framework tuner information. Refer to the Android Media TV Tuner API for details.

Constant Value: 1 (0x00000001)

ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT

Added in API level 34

public static final int ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT

Encapsulation metadata type for placement of supplementary audio. A 32 bit integer constant, one of SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL, SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT, SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT.

Constant Value: 3 (0x00000003)

ENCAPSULATION_MODE_ELEMENTARY_STREAM

Added in API level 30

public static final int ENCAPSULATION_MODE_ELEMENTARY_STREAM

This mode indicates metadata encapsulation with an elementary stream payload. Both compressed and PCM format is allowed.

Constant Value: 1 (0x00000001)

ENCAPSULATION_MODE_NONE

Added in API level 30

public static final int ENCAPSULATION_MODE_NONE

This mode indicates no metadata encapsulation, which is the default mode for sending audio data through AudioTrack.

Constant Value: 0 (0x00000000)

ERROR

Added in API level 3

public static final int ERROR

Denotes a generic operation failure.

Constant Value: -1 (0xffffffff)

ERROR_BAD_VALUE

Added in API level 3

public static final int ERROR_BAD_VALUE

Denotes a failure due to the use of an invalid value.

Constant Value: -2 (0xfffffffe)

ERROR_DEAD_OBJECT

Added in API level 24

public static final int ERROR_DEAD_OBJECT

An error code indicating that the object reporting it is no longer valid and needs to be recreated.

Constant Value: -6 (0xfffffffa)

ERROR_INVALID_OPERATION

Added in API level 3

public static final int ERROR_INVALID_OPERATION

Denotes a failure due to the improper use of a method.

Constant Value: -3 (0xfffffffd)

MODE_STATIC

Added in API level 3

public static final int MODE_STATIC

Creation mode where audio data is transferred from Java to the native layer only once before the audio starts playing.

Constant Value: 0 (0x00000000)

MODE_STREAM

Added in API level 3

public static final int MODE_STREAM

Creation mode where audio data is streamed from Java to the native layer as the audio is playing.

Constant Value: 1 (0x00000001)

PERFORMANCE_MODE_LOW_LATENCY

Added in API level 26

public static final int PERFORMANCE_MODE_LOW_LATENCY

Low latency performance mode for an AudioTrack. If the device supports it, this mode enables a lower latency path through to the audio output sink. Effects may no longer work with such an AudioTrack and the sample rate must match that of the output sink.

Applications should be aware that low latency requires careful buffer management, with smaller chunks of audio data written by each write() call.

If this flag is used without specifying a bufferSizeInBytes then the AudioTrack's actual buffer size may be too small. It is recommended that a fairly large buffer should be specified when the AudioTrack is created. Then the actual size can be reduced by calling setBufferSizeInFrames(int). The buffer size can be optimized by lowering it after each write() call until the audio glitches, which is detected by calling getUnderrunCount(). Then the buffer size can be increased until there are no glitches. This tuning step should be done while playing silence. This technique provides a compromise between latency and glitch rate.

Constant Value: 1 (0x00000001)

PERFORMANCE_MODE_NONE

Added in API level 26

public static final int PERFORMANCE_MODE_NONE

Default performance mode for an AudioTrack.

Constant Value: 0 (0x00000000)

PERFORMANCE_MODE_POWER_SAVING

Added in API level 26

public static final int PERFORMANCE_MODE_POWER_SAVING

Power saving performance mode for an AudioTrack. If the device supports it, this mode will enable a lower power path to the audio output sink. In addition, this lower power path typically will have deeper internal buffers and better underrun resistance, with a tradeoff of higher latency.

In this mode, applications should attempt to use a larger buffer size and deliver larger chunks of audio data per write() call. Use getBufferSizeInFrames() to determine the actual buffer size of the AudioTrack as it may have increased to accommodate a deeper buffer.

Constant Value: 2 (0x00000002)

PLAYSTATE_PAUSED

Added in API level 3

public static final int PLAYSTATE_PAUSED

indicates AudioTrack state is paused

Constant Value: 2 (0x00000002)

PLAYSTATE_PLAYING

Added in API level 3

public static final int PLAYSTATE_PLAYING

indicates AudioTrack state is playing

Constant Value: 3 (0x00000003)

PLAYSTATE_STOPPED

Added in API level 3

public static final int PLAYSTATE_STOPPED

indicates AudioTrack state is stopped

Constant Value: 1 (0x00000001)

STATE_INITIALIZED

Added in API level 3

public static final int STATE_INITIALIZED

State of an AudioTrack that is ready to be used.

Constant Value: 1 (0x00000001)

STATE_NO_STATIC_DATA

Added in API level 3

public static final int STATE_NO_STATIC_DATA

State of a successfully initialized AudioTrack that uses static data, but that hasn't received that data yet.

Constant Value: 2 (0x00000002)

STATE_UNINITIALIZED

Added in API level 3

public static final int STATE_UNINITIALIZED

State of an AudioTrack that was not successfully initialized upon creation.

Constant Value: 0 (0x00000000)

SUCCESS

Added in API level 3

public static final int SUCCESS

Denotes a successful operation.

Constant Value: 0 (0x00000000)

SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT

Added in API level 34

public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT

Supplementary audio placement left.

Constant Value: 1 (0x00000001)

SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL

Added in API level 34

public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL

Supplementary audio placement normal.

Constant Value: 0 (0x00000000)

SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT

Added in API level 34

public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT

Supplementary audio placement right.

Constant Value: 2 (0x00000002)

WRITE_BLOCKING

Added in API level 21

public static final int WRITE_BLOCKING

The write mode indicating the write operation will block until all data has been written, to be used as the actual value of the writeMode parameter in write(byte[], int, int, int), write(short[], int, int, int), write(float[], int, int, int), write(java.nio.ByteBuffer, int, int), and write(java.nio.ByteBuffer, int, int, long).

Constant Value: 0 (0x00000000)

WRITE_NON_BLOCKING

Added in API level 21

public static final int WRITE_NON_BLOCKING

The write mode indicating the write operation will return immediately after queuing as much audio data for playback as possible without blocking, to be used as the actual value of the writeMode parameter in write(java.nio.ByteBuffer, int, int), write(short[], int, int, int), write(float[], int, int, int), write(java.nio.ByteBuffer, int, int), and write(java.nio.ByteBuffer, int, int, long).

Constant Value: 1 (0x00000001)

Public constructors

AudioTrack

Added in API level 3

public AudioTrack (int streamType, 
                int sampleRateInHz, 
                int channelConfig, 
                int audioFormat, 
                int bufferSizeInBytes, 
                int mode)

This constructor is deprecated.
use Builder or AudioTrack(android.media.AudioAttributes, android.media.AudioFormat, int, int, int) to specify the AudioAttributes instead of the stream type which is only for volume control.

Class constructor.

Parameters
`streamType`	`int`: the type of the audio stream. See `AudioManager#STREAM_VOICE_CALL`, `AudioManager#STREAM_SYSTEM`, `AudioManager#STREAM_RING`, `AudioManager#STREAM_MUSIC`, `AudioManager#STREAM_ALARM`, and `AudioManager#STREAM_NOTIFICATION`.
`sampleRateInHz`	`int`: the initial source sample rate expressed in Hz. `AudioFormat#SAMPLE_RATE_UNSPECIFIED` means to use a route-dependent value which is usually the sample rate of the sink. `getSampleRate()` can be used to retrieve the actual sample rate chosen.
`channelConfig`	`int`: describes the configuration of the audio channels. See `AudioFormat#CHANNEL_OUT_MONO` and `AudioFormat#CHANNEL_OUT_STEREO`
`audioFormat`	`int`: the format in which the audio data is represented. See `AudioFormat#ENCODING_PCM_16BIT`, `AudioFormat#ENCODING_PCM_8BIT`, and `AudioFormat#ENCODING_PCM_FLOAT`.
`bufferSizeInBytes`	`int`: the total size (in bytes) of the internal buffer where audio data is read from for playback. This should be a nonzero multiple of the frame size in bytes. If the track's creation mode is `MODE_STATIC`, this is the maximum length sample, or audio clip, that can be played by this instance. If the track's creation mode is `MODE_STREAM`, this should be the desired buffer size for the `AudioTrack` to satisfy the application's latency requirements. If `bufferSizeInBytes` is less than the minimum buffer size for the output sink, it is increased to the minimum buffer size. The method `getBufferSizeInFrames()` returns the actual size in frames of the buffer created, which determines the minimum frequency to write to the streaming `AudioTrack` to avoid underrun. See `getMinBufferSize(int, int, int)` to determine the estimated minimum buffer size for an AudioTrack instance in streaming mode.
`mode`	`int`: streaming or static buffer. See `MODE_STATIC` and `MODE_STREAM`

Throws
	java.lang.IllegalArgumentException
`IllegalArgumentException`

AudioTrack

Added in API level 9

public AudioTrack (int streamType, 
                int sampleRateInHz, 
                int channelConfig, 
                int audioFormat, 
                int bufferSizeInBytes, 
                int mode, 
                int sessionId)

Class constructor with audio session. Use this constructor when the AudioTrack must be attached to a particular audio session. The primary use of the audio session ID is to associate audio effects to a particular instance of AudioTrack: if an audio session ID is provided when creating an AudioEffect, this effect will be applied only to audio tracks and media players in the same session and not to the output mix. When an AudioTrack is created without specifying a session, it will create its own session which can be retrieved by calling the getAudioSessionId() method. If a non-zero session ID is provided, this AudioTrack will share effects attached to this session with all other media players or audio tracks in the same session, otherwise a new session will be created for this track if none is supplied.

Parameters
`streamType`	`int`: the type of the audio stream. See `AudioManager#STREAM_VOICE_CALL`, `AudioManager#STREAM_SYSTEM`, `AudioManager#STREAM_RING`, `AudioManager#STREAM_MUSIC`, `AudioManager#STREAM_ALARM`, and `AudioManager#STREAM_NOTIFICATION`.
`sampleRateInHz`	`int`: the initial source sample rate expressed in Hz. `AudioFormat#SAMPLE_RATE_UNSPECIFIED` means to use a route-dependent value which is usually the sample rate of the sink.
`channelConfig`	`int`: describes the configuration of the audio channels. See `AudioFormat#CHANNEL_OUT_MONO` and `AudioFormat#CHANNEL_OUT_STEREO`
`audioFormat`	`int`: the format in which the audio data is represented. See `AudioFormat#ENCODING_PCM_16BIT` and `AudioFormat#ENCODING_PCM_8BIT`, and `AudioFormat#ENCODING_PCM_FLOAT`.
`bufferSizeInBytes`	`int`: the total size (in bytes) of the internal buffer where audio data is read from for playback. This should be a nonzero multiple of the frame size in bytes. If the track's creation mode is `MODE_STATIC`, this is the maximum length sample, or audio clip, that can be played by this instance. If the track's creation mode is `MODE_STREAM`, this should be the desired buffer size for the `AudioTrack` to satisfy the application's latency requirements. If `bufferSizeInBytes` is less than the minimum buffer size for the output sink, it is increased to the minimum buffer size. The method `getBufferSizeInFrames()` returns the actual size in frames of the buffer created, which determines the minimum frequency to write to the streaming `AudioTrack` to avoid underrun. You can write data into this buffer in smaller chunks than this size. See `getMinBufferSize(int, int, int)` to determine the estimated minimum buffer size for an AudioTrack instance in streaming mode.
`mode`	`int`: streaming or static buffer. See `MODE_STATIC` and `MODE_STREAM`
`sessionId`	`int`: Id of audio session the AudioTrack must be attached to

Throws
	java.lang.IllegalArgumentException
`IllegalArgumentException`

AudioTrack

Added in API level 21

public AudioTrack (AudioAttributes attributes, 
                AudioFormat format, 
                int bufferSizeInBytes, 
                int mode, 
                int sessionId)

Class constructor with AudioAttributes and AudioFormat.

Parameters
`attributes`	`AudioAttributes`: a non-null `AudioAttributes` instance.
`format`	`AudioFormat`: a non-null `AudioFormat` instance describing the format of the data that will be played through this AudioTrack. See `AudioFormat.Builder` for configuring the audio format parameters such as encoding, channel mask and sample rate.
`bufferSizeInBytes`	`int`: the total size (in bytes) of the internal buffer where audio data is read from for playback. This should be a nonzero multiple of the frame size in bytes. If the track's creation mode is `MODE_STATIC`, this is the maximum length sample, or audio clip, that can be played by this instance. If the track's creation mode is `MODE_STREAM`, this should be the desired buffer size for the `AudioTrack` to satisfy the application's latency requirements. If `bufferSizeInBytes` is less than the minimum buffer size for the output sink, it is increased to the minimum buffer size. The method `getBufferSizeInFrames()` returns the actual size in frames of the buffer created, which determines the minimum frequency to write to the streaming `AudioTrack` to avoid underrun. See `getMinBufferSize(int, int, int)` to determine the estimated minimum buffer size for an AudioTrack instance in streaming mode.
`mode`	`int`: streaming or static buffer. See `MODE_STATIC` and `MODE_STREAM`.
`sessionId`	`int`: ID of audio session the AudioTrack must be attached to, or `AudioManager#AUDIO_SESSION_ID_GENERATE` if the session isn't known at construction time. See also `AudioManager#generateAudioSessionId()` to obtain a session ID before construction.

Throws
	java.lang.IllegalArgumentException
`IllegalArgumentException`

Public methods

addOnCodecFormatChangedListener

Added in API level 30

public void addOnCodecFormatChangedListener (Executor executor, 
                AudioTrack.OnCodecFormatChangedListener listener)

Adds an OnCodecFormatChangedListener to receive notifications of codec format change events on this AudioTrack.

Parameters
`executor`	`Executor`: Specifies the `Executor` object to control execution. This value cannot be `null`. Callback and listener events are dispatched through this `Executor`, providing an easy way to control which thread is used. To dispatch events through the main thread of your application, you can use `Context.getMainExecutor()`. Otherwise, provide an `Executor` that dispatches to an appropriate thread.
`listener`	`AudioTrack.OnCodecFormatChangedListener`: The `OnCodecFormatChangedListener` interface to receive notifications of codec events. This value cannot be `null`.

addOnRoutingChangedListener

Added in API level 23
Deprecated in API level 24

public void addOnRoutingChangedListener (AudioTrack.OnRoutingChangedListener listener, 
                Handler handler)

This method was deprecated in API level 24.
users should switch to the general purpose AudioRouting.OnRoutingChangedListener class instead.

Adds an OnRoutingChangedListener to receive notifications of routing changes on this AudioTrack.

Parameters
`listener`	`AudioTrack.OnRoutingChangedListener`: The `OnRoutingChangedListener` interface to receive notifications of rerouting events.
`handler`	`Handler`: Specifies the `Handler` object for the thread on which to execute the callback. If `null`, the `Handler` associated with the main `Looper` will be used.

addOnRoutingChangedListener

Added in API level 24

public void addOnRoutingChangedListener (AudioRouting.OnRoutingChangedListener listener, 
                Handler handler)

Adds an AudioRouting.OnRoutingChangedListener to receive notifications of routing changes on this AudioTrack.

Parameters
`listener`	`AudioRouting.OnRoutingChangedListener`: The `AudioRouting.OnRoutingChangedListener` interface to receive notifications of rerouting events.
`handler`	`Handler`: Specifies the `Handler` object for the thread on which to execute the callback. If `null`, the `Handler` associated with the main `Looper` will be used.

attachAuxEffect

Added in API level 9

public int attachAuxEffect (int effectId)

Attaches an auxiliary effect to the audio track. A typical auxiliary effect is a reverberation effect which can be applied on any sound source that directs a certain amount of its energy to this effect. This amount is defined by setAuxEffectSendLevel(). .

After creating an auxiliary effect (e.g. EnvironmentalReverb), retrieve its ID with AudioEffect.getId() and use it when calling this method to attach the audio track to the effect.

To detach the effect from the audio track, call this method with a null effect id.

Parameters
`effectId`	`int`: system wide unique id of the effect to attach

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_INVALID_OPERATION`, `ERROR_BAD_VALUE`

See also:

setAuxEffectSendLevel(float)

createVolumeShaper

Added in API level 26

public VolumeShaper createVolumeShaper (VolumeShaper.Configuration configuration)

Returns a VolumeShaper object that can be used modify the volume envelope of the player or track.

Parameters
`configuration`	`VolumeShaper.Configuration`: This value cannot be `null`.

Returns
`VolumeShaper`	This value cannot be `null`.

flush

Added in API level 3

public void flush ()

Flushes the audio data currently queued for playback. Any data that has been written but not yet presented will be discarded. No-op if not stopped or paused, or if the track's creation mode is not MODE_STREAM.
Note that although data written but not yet presented is discarded, there is no guarantee that all of the buffer space formerly used by that data is available for a subsequent write. For example, a call to write(byte[], int, int) with sizeInBytes less than or equal to the total buffer size may return a short actual transfer count.

getAudioAttributes

Added in API level 29

public AudioAttributes getAudioAttributes ()

Returns the AudioAttributes used in configuration. If a streamType is used instead of an AudioAttributes to configure the AudioTrack (the use of streamType for configuration is deprecated), then the AudioAttributes equivalent to the streamType is returned.

Returns
`AudioAttributes`	The `AudioAttributes` used to configure the AudioTrack. This value cannot be `null`.

Throws
`IllegalStateException`	If the track is not initialized.

getAudioDescriptionMixLeveldB

Added in API level 30

public float getAudioDescriptionMixLeveldB ()

Returns the Audio Description mix level in dB. If Audio Description mixing is unavailable from the hardware device, a value of Float.NEGATIVE_INFINITY is returned.

Returns
`float`	the current Audio Description Mix Level in dB. A value of `Float.NEGATIVE_INFINITY` means that the audio description is not mixed or the hardware is not available. This should reflect the true internal device mix level; hence the application might receive any floating value except `Float.NaN`.

getAudioFormat

Added in API level 3

public int getAudioFormat ()

Returns the configured audio data encoding. See AudioFormat#ENCODING_PCM_8BIT, AudioFormat#ENCODING_PCM_16BIT, and AudioFormat#ENCODING_PCM_FLOAT.

Returns
`int`

getAudioSessionId

Added in API level 9

public int getAudioSessionId ()

Returns the audio session ID.

Returns
`int`	the ID of the audio session this AudioTrack belongs to.

getBufferCapacityInFrames

Added in API level 24

public int getBufferCapacityInFrames ()

Returns the maximum size of the AudioTrack buffer in frames.

If the track's creation mode is MODE_STATIC, it is equal to the specified bufferSizeInBytes on construction, converted to frame units. A static track's frame count will not change.

If the track's creation mode is MODE_STREAM, it is greater than or equal to the specified bufferSizeInBytes converted to frame units. For streaming tracks, this value may be rounded up to a larger value if needed by the target output sink, and if the track is subsequently routed to a different output sink, the frame count may enlarge to accommodate.

If the AudioTrack encoding indicates compressed data, e.g. AudioFormat#ENCODING_AC3, then the frame count returned is the size of the AudioTrack buffer in bytes.

Returns
`int`	maximum size in frames of the `AudioTrack` buffer. Value is 0 or greater

Throws
`IllegalStateException`	if track is not initialized.

getBufferSizeInFrames

Added in API level 23

public int getBufferSizeInFrames ()

Returns the effective size of the AudioTrack buffer that the application writes to.

This will be less than or equal to the result of getBufferCapacityInFrames(). It will be equal if setBufferSizeInFrames(int) has never been called.

If the track is subsequently routed to a different output sink, the buffer size and capacity may enlarge to accommodate.

If the AudioTrack encoding indicates compressed data, e.g. AudioFormat#ENCODING_AC3, then the frame count returned is the size of the AudioTrack buffer in bytes.

Returns
`int`	current size in frames of the `AudioTrack` buffer. Value is 0 or greater

Throws
`IllegalStateException`	if track is not initialized.

getChannelConfiguration

Added in API level 3

public int getChannelConfiguration ()

Returns the configured channel position mask.

For example, refer to AudioFormat#CHANNEL_OUT_MONO, AudioFormat#CHANNEL_OUT_STEREO, AudioFormat#CHANNEL_OUT_5POINT1. This method may return AudioFormat#CHANNEL_INVALID if a channel index mask was used. Consider getFormat() instead, to obtain an AudioFormat, which contains both the channel position mask and the channel index mask.

Returns
`int`

getChannelCount

Added in API level 3

public int getChannelCount ()

Returns the configured number of channels.

Returns
`int`

getDualMonoMode

Added in API level 30

public int getDualMonoMode ()

Returns the Dual Mono mode presentation setting. If no Dual Mono presentation is available for the output device, then DUAL_MONO_MODE_OFF is returned.

Returns
`int`	one of `DUAL_MONO_MODE_OFF`, `DUAL_MONO_MODE_LR`, `DUAL_MONO_MODE_LL`, `DUAL_MONO_MODE_RR`. Value is `DUAL_MONO_MODE_OFF`, `DUAL_MONO_MODE_LR`, `DUAL_MONO_MODE_LL`, or `DUAL_MONO_MODE_RR`

getFormat

Added in API level 23

public AudioFormat getFormat ()

Returns the configured AudioTrack format.

Returns
`AudioFormat`	an `AudioFormat` containing the `AudioTrack` parameters at the time of configuration. This value cannot be `null`.

getLogSessionId

Added in API level 31

public LogSessionId getLogSessionId ()

Returns the LogSessionId.

Returns
`LogSessionId`	This value cannot be `null`.

getMaxVolume

Added in API level 3

public static float getMaxVolume ()

Returns the maximum gain value, which is greater than or equal to 1.0. Gain values greater than the maximum will be clamped to the maximum.

The word "volume" in the API name is historical; this is actually a gain. expressed as a linear multiplier on sample values, where a maximum value of 1.0 corresponds to a gain of 0 dB (sample values left unmodified).

Returns
`float`	the maximum value, which is greater than or equal to 1.0.

getMetrics

Added in API level 28

public PersistableBundle getMetrics ()

Return Metrics data about the current AudioTrack instance.

Returns
`PersistableBundle`	a `PersistableBundle` containing the set of attributes and values available for the media being handled by this instance of AudioTrack The attributes are descibed in `MetricsConstants`. Additional vendor-specific fields may also be present in the return value.

getMinBufferSize

Added in API level 3

public static int getMinBufferSize (int sampleRateInHz, 
                int channelConfig, 
                int audioFormat)

Returns the estimated minimum buffer size required for an AudioTrack object to be created in the MODE_STREAM mode. The size is an estimate because it does not consider either the route or the sink, since neither is known yet. Note that this size doesn't guarantee a smooth playback under load, and higher values should be chosen according to the expected frequency at which the buffer will be refilled with additional data to play. For example, if you intend to dynamically set the source sample rate of an AudioTrack to a higher value than the initial source sample rate, be sure to configure the buffer size based on the highest planned sample rate.

Parameters
`sampleRateInHz`	`int`: the source sample rate expressed in Hz. `AudioFormat#SAMPLE_RATE_UNSPECIFIED` is not permitted.
`channelConfig`	`int`: describes the configuration of the audio channels. See `AudioFormat#CHANNEL_OUT_MONO` and `AudioFormat#CHANNEL_OUT_STEREO`
`audioFormat`	`int`: the format in which the audio data is represented. See `AudioFormat#ENCODING_PCM_16BIT` and `AudioFormat#ENCODING_PCM_8BIT`, and `AudioFormat#ENCODING_PCM_FLOAT`.

Returns
`int`	`ERROR_BAD_VALUE` if an invalid parameter was passed, or `ERROR` if unable to query for output properties, or the minimum buffer size expressed in bytes.

getMinVolume

Added in API level 3

public static float getMinVolume ()

Returns the minimum gain value, which is the constant 0.0. Gain values less than 0.0 will be clamped to 0.0.

The word "volume" in the API name is historical; this is actually a linear gain.

Returns
`float`	the minimum value, which is the constant 0.0.

getNativeOutputSampleRate

Added in API level 3

public static int getNativeOutputSampleRate (int streamType)

Returns the output sample rate in Hz for the specified stream type.

Parameters
`streamType`	`int`

Returns
`int`

getNotificationMarkerPosition

Added in API level 3

public int getNotificationMarkerPosition ()

Returns marker position expressed in frames.

Returns
`int`	marker position in wrapping frame units similar to `getPlaybackHeadPosition()`, or zero if marker is disabled.

getOffloadDelay

Added in API level 29

public int getOffloadDelay ()

Return the decoder delay of an offloaded track, expressed in frames, previously set with setOffloadDelayPadding(int, int), or 0 if it was never modified.

This delay indicates the number of frames to be ignored at the beginning of the stream. This value can only be queried on a track successfully initialized with AudioTrack.Builder#setOffloadedPlayback(boolean).

Returns
`int`	decoder delay expressed in frames. Value is 0 or greater

getOffloadPadding

Added in API level 29

public int getOffloadPadding ()

Return the decoder padding of an offloaded track, expressed in frames, previously set with setOffloadDelayPadding(int, int), or 0 if it was never modified.

This padding indicates the number of frames to be ignored at the end of the stream. This value can only be queried on a track successfully initialized with AudioTrack.Builder#setOffloadedPlayback(boolean).

Returns
`int`	decoder padding expressed in frames. Value is 0 or greater

getPerformanceMode

Added in API level 26

public int getPerformanceMode ()

Returns the current performance mode of the AudioTrack.

Returns
`int`	one of `AudioTrack#PERFORMANCE_MODE_NONE`, `AudioTrack#PERFORMANCE_MODE_LOW_LATENCY`, or `AudioTrack#PERFORMANCE_MODE_POWER_SAVING`. Use `AudioTrack.Builder#setPerformanceMode` in the `AudioTrack.Builder` to enable a performance mode. Value is `PERFORMANCE_MODE_NONE`, `PERFORMANCE_MODE_LOW_LATENCY`, or `PERFORMANCE_MODE_POWER_SAVING`

Throws
`IllegalStateException`	if track is not initialized.

getPlayState

Added in API level 3

public int getPlayState ()

Returns the playback state of the AudioTrack instance.

Returns
`int`

See also:

getPlaybackHeadPosition

Added in API level 3

public int getPlaybackHeadPosition ()

Returns the playback head position expressed in frames. Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. This is a continuously advancing counter. It will wrap (overflow) periodically, for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. It is reset to zero by flush(), reloadStaticData(), and stop(). If the track's creation mode is MODE_STATIC, the return value indicates the total number of frames played since reset, not the current offset within the buffer.

Returns
`int`

getPlaybackParams

Added in API level 23

public PlaybackParams getPlaybackParams ()

Returns the current playback parameters. See setPlaybackParams(android.media.PlaybackParams) to set playback parameters

Returns
`PlaybackParams`	current `PlaybackParams`. This value cannot be `null`.

Throws
`IllegalStateException`	if track is not initialized.

getPlaybackRate

Added in API level 3

public int getPlaybackRate ()

Returns the current playback sample rate rate in Hz.

Returns
`int`

getPositionNotificationPeriod

Added in API level 3

public int getPositionNotificationPeriod ()

Returns the notification update period expressed in frames. Zero means that no position update notifications are being delivered.

Returns
`int`

getPreferredDevice

Added in API level 23

public AudioDeviceInfo getPreferredDevice ()

Returns the selected output specified by setPreferredDevice(AudioDeviceInfo). Note that this is not guaranteed to correspond to the actual device being used for playback.

Returns
`AudioDeviceInfo`

getRoutedDevice

Added in API level 23

public AudioDeviceInfo getRoutedDevice ()

Returns an AudioDeviceInfo identifying the current routing of this AudioTrack. Note: The query is only valid if the AudioTrack is currently playing. If it is not, getRoutedDevice() will return null.

Returns
`AudioDeviceInfo`

getSampleRate

Added in API level 3

public int getSampleRate ()

Returns the configured audio source sample rate in Hz. The initial source sample rate depends on the constructor parameters, but the source sample rate may change if setPlaybackRate(int) is called. If the constructor had a specific sample rate, then the initial sink sample rate is that value. If the constructor had AudioFormat#SAMPLE_RATE_UNSPECIFIED, then the initial sink sample rate is a route-dependent default value based on the source [sic].

Returns
`int`

getStartThresholdInFrames

Added in API level 31

public int getStartThresholdInFrames ()

Returns the streaming start threshold of the AudioTrack.

The streaming start threshold is the buffer level that the written audio data must reach for audio streaming to start after play() is called. When an AudioTrack is created, the streaming start threshold is the buffer capacity in frames. If the buffer size in frames is reduced by setBufferSizeInFrames(int) to a value smaller than the start threshold then that value will be used instead for the streaming start threshold.

For compressed streams, the size of a frame is considered to be exactly one byte.

Returns
`int`	the current start threshold in frames value. This is an integer between 1 to the buffer capacity (see `getBufferCapacityInFrames()`), and might change if the output sink changes after track creation. Value is 1 or greater

Throws
`IllegalStateException`	if the track is not initialized or the track is not `MODE_STREAM`.

See also:

setStartThresholdInFrames(int)

getState

Added in API level 3

public int getState ()

Returns the state of the AudioTrack instance. This is useful after the AudioTrack instance has been created to check if it was initialized properly. This ensures that the appropriate resources have been acquired.

Returns
`int`

See also:

getStreamType

Added in API level 3

public int getStreamType ()

Returns the volume stream type of this AudioTrack. Compare the result against AudioManager#STREAM_VOICE_CALL, AudioManager#STREAM_SYSTEM, AudioManager#STREAM_RING, AudioManager#STREAM_MUSIC, AudioManager#STREAM_ALARM, AudioManager#STREAM_NOTIFICATION, AudioManager#STREAM_DTMF or AudioManager#STREAM_ACCESSIBILITY.

Returns
`int`

getTimestamp

Added in API level 19

public boolean getTimestamp (AudioTimestamp timestamp)

Poll for a timestamp on demand.

If you need to track timestamps during initial warmup or after a routing or mode change, you should request a new timestamp periodically until the reported timestamps show that the frame position is advancing, or until it becomes clear that timestamps are unavailable for this route.

After the clock is advancing at a stable rate, query for a new timestamp approximately once every 10 seconds to once per minute. Calling this method more often is inefficient. It is also counter-productive to call this method more often than recommended, because the short-term differences between successive timestamp reports are not meaningful. If you need a high-resolution mapping between frame position and presentation time, consider implementing that at application level, based on low-resolution timestamps.

The audio data at the returned position may either already have been presented, or may have not yet been presented but is committed to be presented. It is not possible to request the time corresponding to a particular position, or to request the (fractional) position corresponding to a particular time. If you need such features, consider implementing them at application level.

Parameters
`timestamp`	`AudioTimestamp`: a reference to a non-null AudioTimestamp instance allocated and owned by caller.

Returns

boolean

true if a timestamp is available, or false if no timestamp is available. If a timestamp is available, the AudioTimestamp instance is filled in with a position in frame units, together with the estimated time when that frame was presented or is committed to be presented. In the case that no timestamp is available, any supplied instance is left unaltered. A timestamp may be temporarily unavailable while the audio clock is stabilizing, or during and immediately after a route change. A timestamp is permanently unavailable for a given route if the route does not support timestamps. In this case, the approximate frame position can be obtained using getPlaybackHeadPosition(). However, it may be useful to continue to query for timestamps occasionally, to recover after a route change.

getUnderrunCount

Added in API level 24

public int getUnderrunCount ()

Returns the number of underrun occurrences in the application-level write buffer since the AudioTrack was created. An underrun occurs if the application does not write audio data quickly enough, causing the buffer to underflow and a potential audio glitch or pop.

Underruns are less likely when buffer sizes are large. It may be possible to eliminate underruns by recreating the AudioTrack with a larger buffer. Or by using setBufferSizeInFrames(int) to dynamically increase the effective size of the buffer.

Returns
`int`

isDirectPlaybackSupported

Added in API level 29
Deprecated in API level 33

public static boolean isDirectPlaybackSupported (AudioFormat format, 
                AudioAttributes attributes)

This method was deprecated in API level 33.
Use AudioManager#getDirectPlaybackSupport(AudioFormat, AudioAttributes) instead.

Returns whether direct playback of an audio format with the provided attributes is currently supported on the system.

Direct playback means that the audio stream is not resampled or downmixed by the framework. Checking for direct support can help the app select the representation of audio content that most closely matches the capabilities of the device and peripherials (e.g. A/V receiver) connected to it. Note that the provided stream can still be re-encoded or mixed with other streams, if needed.

Also note that this query only provides information about the support of an audio format. It does not indicate whether the resources necessary for the playback are available at that instant.

Parameters
`format`	`AudioFormat`: a non-null `AudioFormat` instance describing the format of the audio data.
`attributes`	`AudioAttributes`: a non-null `AudioAttributes` instance.

Returns
`boolean`	true if the given audio format can be played directly.

isOffloadedPlayback

Added in API level 29

public boolean isOffloadedPlayback ()

Returns whether the track was built with Builder#setOffloadedPlayback(boolean) set to true.

Returns
`boolean`	true if the track is using offloaded playback.

pause

Added in API level 3

public void pause ()

Pauses the playback of the audio data. Data that has not been played back will not be discarded. Subsequent calls to play() will play this data back. See flush() to discard this data.

Throws
	java.lang.IllegalStateException
`IllegalStateException`

play

Added in API level 3

public void play ()

Starts playing an AudioTrack.

If track's creation mode is MODE_STATIC, you must have called one of the write methods (write(byte[], int, int), write(byte[], int, int, int), write(short[], int, int), write(short[], int, int, int), write(float[], int, int, int), or write(java.nio.ByteBuffer, int, int)) prior to play().

If the mode is MODE_STREAM, you can optionally prime the data path prior to calling play(), by writing up to bufferSizeInBytes (from constructor). If you don't call write() first, or if you call write() but with an insufficient amount of data, then the track will be in underrun state at play(). In this case, playback will not actually start playing until the data path is filled to a device-specific minimum level. This requirement for the path to be filled to a minimum level is also true when resuming audio playback after calling stop(). Similarly the buffer will need to be filled up again after the track underruns due to failure to call write() in a timely manner with sufficient data. For portability, an application should prime the data path to the maximum allowed by writing data until the write() method returns a short transfer count. This allows play() to start immediately, and reduces the chance of underrun.

As of Build.VERSION_CODES.S the minimum level to start playing can be obtained using getStartThresholdInFrames() and set with setStartThresholdInFrames(int).

Throws
`IllegalStateException`	if the track isn't properly initialized

registerStreamEventCallback

Added in API level 29

public void registerStreamEventCallback (Executor executor, 
                AudioTrack.StreamEventCallback eventCallback)

Registers a callback for the notification of stream events. This callback can only be registered for instances operating in offloaded mode (see AudioTrack.Builder#setOffloadedPlayback(boolean) and AudioManager#isOffloadedPlaybackSupported(AudioFormat,AudioAttributes) for more details).

Parameters
`executor`	`Executor`: `Executor` to handle the callbacks. This value cannot be `null`. Callback and listener events are dispatched through this `Executor`, providing an easy way to control which thread is used. To dispatch events through the main thread of your application, you can use `Context.getMainExecutor()`. Otherwise, provide an `Executor` that dispatches to an appropriate thread.
`eventCallback`	`AudioTrack.StreamEventCallback`: the callback to receive the stream event notifications. This value cannot be `null`.

release

Added in API level 3

public void release ()

Releases the native AudioTrack resources.

reloadStaticData

Added in API level 3

public int reloadStaticData ()

Sets the playback head position within the static buffer to zero, that is it rewinds to start of static buffer. The track must be stopped or paused, and the track's creation mode must be MODE_STATIC.

As of Build.VERSION_CODES.M, also resets the value returned by getPlaybackHeadPosition() to zero. For earlier API levels, the reset behavior is unspecified.

Use setPlaybackHeadPosition(int) with a zero position if the reset of getPlaybackHeadPosition() is not needed.

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

removeOnCodecFormatChangedListener

Added in API level 30

public void removeOnCodecFormatChangedListener (AudioTrack.OnCodecFormatChangedListener listener)

Removes an OnCodecFormatChangedListener which has been previously added to receive codec format change events.

Parameters
`listener`	`AudioTrack.OnCodecFormatChangedListener`: The previously added `OnCodecFormatChangedListener` interface to remove. This value cannot be `null`.

removeOnRoutingChangedListener

Added in API level 23
Deprecated in API level 24

public void removeOnRoutingChangedListener (AudioTrack.OnRoutingChangedListener listener)

This method was deprecated in API level 24.
users should switch to the general purpose AudioRouting.OnRoutingChangedListener class instead.

Removes an OnRoutingChangedListener which has been previously added to receive rerouting notifications.

Parameters
`listener`	`AudioTrack.OnRoutingChangedListener`: The previously added `OnRoutingChangedListener` interface to remove.

removeOnRoutingChangedListener

Added in API level 24

public void removeOnRoutingChangedListener (AudioRouting.OnRoutingChangedListener listener)

Removes an AudioRouting.OnRoutingChangedListener which has been previously added to receive rerouting notifications.

Parameters
`listener`	`AudioRouting.OnRoutingChangedListener`: The previously added `AudioRouting.OnRoutingChangedListener` interface to remove.

setAudioDescriptionMixLeveldB

Added in API level 30

public boolean setAudioDescriptionMixLeveldB (float level)

Sets the Audio Description mix level in dB. For AudioTracks incorporating a secondary Audio Description stream (where such contents may be sent through an Encapsulation Mode other than ENCAPSULATION_MODE_NONE). or internally by a HW channel), the level of mixing of the Audio Description to the Main Audio stream is controlled by this method. Such mixing occurs prior to overall volume scaling.

Parameters
`level`	`float`: a floating point value between `Float.NEGATIVE_INFINITY` to `+48.f`, where `Float.NEGATIVE_INFINITY` means the Audio Description is not mixed and a level of `0.f` means the Audio Description is mixed without scaling. Value is 48.f or less

Returns
`boolean`	true on success, false on failure.

setAuxEffectSendLevel

Added in API level 9

public int setAuxEffectSendLevel (float level)

Sets the send level of the audio track to the attached auxiliary effect attachAuxEffect(int). Effect levels are clamped to the closed interval [0.0, max] where max is the value of getMaxVolume(). A value of 0.0 results in no effect, and a value of 1.0 is full send.

By default the send level is 0.0f, so even if an effect is attached to the player this method must be called for the effect to be applied.

Note that the passed level value is a linear scalar. UI controls should be scaled logarithmically: the gain applied by audio framework ranges from -72dB to at least 0dB, so an appropriate conversion from linear UI input x to level is: x == 0 -> level = 0 0 < x <= R -> level = 10^(72*(x-R)/20/R)

Parameters
`level`	`float`: linear send level Value is 0.0 or greater

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_INVALID_OPERATION`, `ERROR`

setBufferSizeInFrames

Added in API level 24

public int setBufferSizeInFrames (int bufferSizeInFrames)

Limits the effective size of the AudioTrack buffer that the application writes to.

A write to this AudioTrack will not fill the buffer beyond this limit. If a blocking write is used then the write will block until the data can fit within this limit.

Changing this limit modifies the latency associated with the buffer for this track. A smaller size will give lower latency but there may be more glitches due to buffer underruns.

The actual size used may not be equal to this requested size. It will be limited to a valid range with a maximum of getBufferCapacityInFrames(). It may also be adjusted slightly for internal reasons. If bufferSizeInFrames is less than zero then ERROR_BAD_VALUE will be returned.

This method is supported for PCM audio at all API levels. Compressed audio is supported in API levels 33 and above. For compressed streams the size of a frame is considered to be exactly one byte.

Parameters
`bufferSizeInFrames`	`int`: requested buffer size in frames Value is 0 or greater

Returns
`int`	the actual buffer size in frames or an error code, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

Throws
`IllegalStateException`	if track is not initialized.

setDualMonoMode

Added in API level 30

public boolean setDualMonoMode (int dualMonoMode)

Sets the Dual Mono mode presentation on the output device. The Dual Mono mode is generally applied to stereo audio streams where the left and right channels come from separate sources. For compressed audio, where the decoding is done in hardware, Dual Mono presentation needs to be performed by the hardware output device as the PCM audio is not available to the framework.

Parameters
`dualMonoMode`	`int`: one of `DUAL_MONO_MODE_OFF`, `DUAL_MONO_MODE_LR`, `DUAL_MONO_MODE_LL`, `DUAL_MONO_MODE_RR`. Value is `DUAL_MONO_MODE_OFF`, `DUAL_MONO_MODE_LR`, `DUAL_MONO_MODE_LL`, or `DUAL_MONO_MODE_RR`

Returns
`boolean`	true on success, false on failure if the output device does not support Dual Mono mode.

setLogSessionId

Added in API level 31

public void setLogSessionId (LogSessionId logSessionId)

Sets a LogSessionId instance to this AudioTrack for metrics collection.

Parameters
`logSessionId`	`LogSessionId`: a `LogSessionId` instance which is used to identify this object to the metrics service. Proper generated Ids must be obtained from the Java metrics service and should be considered opaque. Use `LogSessionId#LOG_SESSION_ID_NONE` to remove the logSessionId association. This value cannot be `null`.

Throws
`IllegalStateException`	if AudioTrack not initialized.

setLoopPoints

Added in API level 3

public int setLoopPoints (int startInFrames, 
                int endInFrames, 
                int loopCount)

Sets the loop points and the loop count. The loop can be infinite. Similarly to setPlaybackHeadPosition, the track must be stopped or paused for the loop points to be changed, and must use the MODE_STATIC mode.

Parameters
`startInFrames`	`int`: loop start marker expressed in frames. Zero corresponds to start of buffer. The start marker must not be greater than or equal to the buffer size in frames, or negative. Value is 0 or greater
`endInFrames`	`int`: loop end marker expressed in frames. The total buffer size in frames corresponds to end of buffer. The end marker must not be greater than the buffer size in frames. For looping, the end marker must not be less than or equal to the start marker, but to disable looping it is permitted for start marker, end marker, and loop count to all be 0. If any input parameters are out of range, this method returns `ERROR_BAD_VALUE`. If the loop period (endInFrames - startInFrames) is too small for the implementation to support, `ERROR_BAD_VALUE` is returned. The loop range is the interval [startInFrames, endInFrames). As of `Build.VERSION_CODES.M`, the position is left unchanged, unless it is greater than or equal to the loop end marker, in which case it is forced to the loop start marker. For earlier API levels, the effect on position is unspecified. Value is 0 or greater
`loopCount`	`int`: the number of times the loop is looped; must be greater than or equal to -1. A value of -1 means infinite looping, and 0 disables looping. A value of positive N means to "loop" (go back) N times. For example, a value of one means to play the region two times in total. Value is -1 or greater

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

setNotificationMarkerPosition

Added in API level 3

public int setNotificationMarkerPosition (int markerInFrames)

Sets the position of the notification marker. At most one marker can be active.

Parameters
`markerInFrames`	`int`: marker position in wrapping frame units similar to `getPlaybackHeadPosition()`, or zero to disable the marker. To set a marker at a position which would appear as zero due to wraparound, a workaround is to use a non-zero position near zero, such as -1 or 1.

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

setOffloadDelayPadding

Added in API level 29

public void setOffloadDelayPadding (int delayInFrames, 
                int paddingInFrames)

Configures the delay and padding values for the current compressed stream playing in offload mode. This can only be used on a track successfully initialized with AudioTrack.Builder#setOffloadedPlayback(boolean). The unit is frames, where a frame indicates the number of samples per channel, e.g. 100 frames for a stereo compressed stream corresponds to 200 decoded interleaved PCM samples.

Parameters
`delayInFrames`	`int`: number of frames to be ignored at the beginning of the stream. A value of 0 indicates no delay is to be applied. Value is 0 or greater
`paddingInFrames`	`int`: number of frames to be ignored at the end of the stream. A value of 0 of 0 indicates no padding is to be applied. Value is 0 or greater

setOffloadEndOfStream

Added in API level 29

public void setOffloadEndOfStream ()

Declares that the last write() operation on this track provided the last buffer of this stream. After the end of stream, previously set padding and delay values are ignored. Can only be called only if the AudioTrack is opened in offload mode . Can only be called only if the AudioTrack is in state PLAYSTATE_PLAYING . Use this method in the same thread as any write() operation.

See also:

setPlaybackHeadPosition

Added in API level 3

public int setPlaybackHeadPosition (int positionInFrames)

Sets the playback head position within the static buffer. The track must be stopped or paused for the position to be changed, and must use the MODE_STATIC mode.

Parameters

positionInFrames

int: playback head position within buffer, expressed in frames. Zero corresponds to start of buffer. The position must not be greater than the buffer size in frames, or negative. Though this method and getPlaybackHeadPosition() have similar names, the position values have different meanings.
If looping is currently enabled and the new position is greater than or equal to the loop end marker, the behavior varies by API level: as of Build.VERSION_CODES.M, the looping is first disabled and then the position is set. For earlier API levels, the behavior is unspecified. Value is 0 or greater

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

setPlaybackParams

Added in API level 23

public void setPlaybackParams (PlaybackParams params)

Sets the playback parameters. This method returns failure if it cannot apply the playback parameters. One possible cause is that the parameters for speed or pitch are out of range. Another possible cause is that the AudioTrack is streaming (see MODE_STREAM) and the buffer size is too small. For speeds greater than 1.0f, the AudioTrack buffer on configuration must be larger than the speed multiplied by the minimum size getMinBufferSize(int, int, int)) to allow proper playback.

Parameters
`params`	`PlaybackParams`: see `PlaybackParams`. In particular, speed, pitch, and audio mode should be set. This value cannot be `null`.

Throws
`IllegalArgumentException`	if the parameters are invalid or not accepted.
`IllegalStateException`	if track is not initialized.

setPlaybackPositionUpdateListener

Added in API level 3

public void setPlaybackPositionUpdateListener (AudioTrack.OnPlaybackPositionUpdateListener listener, 
                Handler handler)

Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update. Use this method to receive AudioTrack events in the Handler associated with another thread than the one in which you created the AudioTrack instance.

Parameters
`handler`	`Handler`: the Handler that will receive the event notification messages.

setPlaybackPositionUpdateListener

Added in API level 3

public void setPlaybackPositionUpdateListener (AudioTrack.OnPlaybackPositionUpdateListener listener)

Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update. Notifications will be received in the same thread as the one in which the AudioTrack instance was created.

setPlaybackRate

Added in API level 3

public int setPlaybackRate (int sampleRateInHz)

Sets the playback sample rate for this track. This sets the sampling rate at which the audio data will be consumed and played back (as set by the sampleRateInHz parameter in the AudioTrack(int, int, int, int, int, int) constructor), not the original sampling rate of the content. For example, setting it to half the sample rate of the content will cause the playback to last twice as long, but will also result in a pitch shift down by one octave. The valid sample rate range is from 1 Hz to twice the value returned by getNativeOutputSampleRate(int). Use setPlaybackParams(android.media.PlaybackParams) for speed control.

This method may also be used to repurpose an existing AudioTrack for playback of content of differing sample rate, but with identical encoding and channel mask.

Parameters
`sampleRateInHz`	`int`: the sample rate expressed in Hz

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

setPositionNotificationPeriod

Added in API level 3

public int setPositionNotificationPeriod (int periodInFrames)

Sets the period for the periodic notification event.

Parameters
`periodInFrames`	`int`: update period expressed in frames. Zero period means no position updates. A negative period is not allowed.

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_INVALID_OPERATION`

setPreferredDevice

Added in API level 23

public boolean setPreferredDevice (AudioDeviceInfo deviceInfo)

Specifies an audio device (via an AudioDeviceInfo object) to route the output from this AudioTrack.

Parameters
`deviceInfo`	`AudioDeviceInfo`: The `AudioDeviceInfo` specifying the audio sink. If deviceInfo is null, default routing is restored.

Returns
`boolean`	true if succesful, false if the specified `AudioDeviceInfo` is non-null and does not correspond to a valid audio output device.

setPresentation

Added in API level 28

public int setPresentation (AudioPresentation presentation)

Sets the audio presentation. If the audio presentation is invalid then ERROR_BAD_VALUE will be returned. If a multi-stream decoder (MSD) is not present, or the format does not support multiple presentations, then ERROR_INVALID_OPERATION will be returned. ERROR is returned in case of any other error.

Parameters
`presentation`	`AudioPresentation`: see `AudioPresentation`. In particular, id should be set. This value cannot be `null`.

Returns
`int`	error code or success, see `SUCCESS`, `ERROR`, `ERROR_BAD_VALUE`, `ERROR_INVALID_OPERATION`

Throws
`IllegalArgumentException`	if the audio presentation is null.
`IllegalStateException`	if track is not initialized.

setStartThresholdInFrames

Added in API level 31

public int setStartThresholdInFrames (int startThresholdInFrames)

Sets the streaming start threshold for an AudioTrack.

The streaming start threshold is the buffer level that the written audio data must reach for audio streaming to start after play() is called.

For compressed streams, the size of a frame is considered to be exactly one byte.

Parameters
`startThresholdInFrames`	`int`: the desired start threshold. Value is 1 or greater

Returns
`int`	the actual start threshold in frames value. This is an integer between 1 to the buffer capacity (see `getBufferCapacityInFrames()`), and might change if the output sink changes after track creation. Value is 1 or greater

Throws
`IllegalStateException`	if the track is not initialized or the track transfer mode is not `MODE_STREAM`.
`IllegalArgumentException`	if startThresholdInFrames is not positive.

See also:

getStartThresholdInFrames()

setStereoVolume

Added in API level 3
Deprecated in API level 21

public int setStereoVolume (float leftGain, 
                float rightGain)

This method was deprecated in API level 21.
Applications should use setVolume(float) instead, as it more gracefully scales down to mono, and up to multi-channel content beyond stereo.

Sets the specified left and right output gain values on the AudioTrack.

Gain values are clamped to the closed interval [0.0, max] where max is the value of getMaxVolume(). A value of 0.0 results in zero gain (silence), and a value of 1.0 means unity gain (signal unchanged). The default value is 1.0 meaning unity gain.

The word "volume" in the API name is historical; this is actually a linear gain.

Parameters
`leftGain`	`float`: output gain for the left channel.
`rightGain`	`float`: output gain for the right channel

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_INVALID_OPERATION`

setVolume

Added in API level 21

public int setVolume (float gain)

Sets the specified output gain value on all channels of this track.

This API is preferred over setStereoVolume(float, float), as it more gracefully scales down to mono, and up to multi-channel content beyond stereo.

The word "volume" in the API name is historical; this is actually a linear gain.

Parameters
`gain`	`float`: output gain for all channels.

Returns
`int`	error code or success, see `SUCCESS`, `ERROR_INVALID_OPERATION`

stop

Added in API level 3

public void stop ()

Stops playing the audio data. When used on an instance created in MODE_STREAM mode, audio will stop playing after the last buffer that was written has been played. For an immediate stop, use pause(), followed by flush() to discard audio data that hasn't been played back yet.

Throws
	java.lang.IllegalStateException
`IllegalStateException`

unregisterStreamEventCallback

Added in API level 29

public void unregisterStreamEventCallback (AudioTrack.StreamEventCallback eventCallback)

Unregisters the callback for notification of stream events, previously registered with registerStreamEventCallback(java.util.concurrent.Executor, android.media.AudioTrack.StreamEventCallback).

Parameters
`eventCallback`	`AudioTrack.StreamEventCallback`: the callback to unregister. This value cannot be `null`.

write

Added in API level 21

public int write (float[] audioData, 
                int offsetInFloats, 
                int sizeInFloats, 
                int writeMode)

Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). The format specified in the AudioTrack constructor should be AudioFormat#ENCODING_PCM_FLOAT to correspond to the data in the array.

In streaming mode, the blocking behavior depends on the write mode. If the write mode is WRITE_BLOCKING, the write will normally block until all the data has been enqueued for playback, and will return a full transfer count. However, if the write mode is WRITE_NON_BLOCKING, or the track is stopped or paused on entry, or another thread interrupts the write by calling stop or pause, or an I/O error occurs during the write, then the write may return a short transfer count.

In static buffer mode, copies the data to the buffer starting at offset 0, and the write mode is ignored. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`float`: the array that holds the data to write. The implementation does not clip for sample values within the nominal range [-1.0f, 1.0f], provided that all gains in the audio pipeline are less than or equal to unity (1.0f), and in the absence of post-processing effects that could add energy, such as reverb. For the convenience of applications that compute samples using filters with non-unity gain, sample values +3 dB beyond the nominal range are permitted. However such values may eventually be limited or clipped, depending on various gains and later processing in the audio path. Therefore applications are encouraged to provide samples values within the nominal range. This value cannot be `null`.
`offsetInFloats`	`int`: the offset, expressed as a number of floats, in audioData where the data to write starts. Must not be negative, or cause the data access to go out of bounds of the array.
`sizeInFloats`	`int`: the number of floats to write in audioData after the offset. Must not be negative, or cause the data access to go out of bounds of the array.
`writeMode`	`int`: one of `WRITE_BLOCKING`, `WRITE_NON_BLOCKING`. It has no effect in static mode. With `WRITE_BLOCKING`, the write will block until all data has been written to the audio sink. With `WRITE_NON_BLOCKING`, the write will return immediately after queuing as much audio data for playback as possible without blocking. Value is `WRITE_BLOCKING`, or `WRITE_NON_BLOCKING`

Returns

int

zero or the positive number of floats that were written, or one of the following error codes. The number of floats will be a multiple of the channel count not to exceed sizeInFloats.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

write

Added in API level 3

public int write (short[] audioData, 
                int offsetInShorts, 
                int sizeInShorts)

In streaming mode, the write will normally block until all the data has been enqueued for playback, and will return a full transfer count. However, if the track is stopped or paused on entry, or another thread interrupts the write by calling stop or pause, or an I/O error occurs during the write, then the write may return a short transfer count.

In static buffer mode, copies the data to the buffer starting at offset 0. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`short`: the array that holds the data to play. This value cannot be `null`.
`offsetInShorts`	`int`: the offset expressed in shorts in audioData where the data to play starts. Must not be negative, or cause the data access to go out of bounds of the array.
`sizeInShorts`	`int`: the number of shorts to read in audioData after the offset. Must not be negative, or cause the data access to go out of bounds of the array.

Returns

int

zero or the positive number of shorts that were written, or one of the following error codes. The number of shorts will be a multiple of the channel count not to exceed sizeInShorts.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

This is equivalent to write(short[], int, int, int) with writeMode set to WRITE_BLOCKING.

write

Added in API level 3

public int write (byte[] audioData, 
                int offsetInBytes, 
                int sizeInBytes)

In static buffer mode, copies the data to the buffer starting at offset 0. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`byte`: the array that holds the data to play. This value cannot be `null`.
`offsetInBytes`	`int`: the offset expressed in bytes in audioData where the data to write starts. Must not be negative, or cause the data access to go out of bounds of the array.
`sizeInBytes`	`int`: the number of bytes to write in audioData after the offset. Must not be negative, or cause the data access to go out of bounds of the array.

Returns

int

zero or the positive number of bytes that were written, or one of the following error codes. The number of bytes will be a multiple of the frame size in bytes not to exceed sizeInBytes.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

This is equivalent to write(byte[], int, int, int) with writeMode set to WRITE_BLOCKING.

write

Added in API level 21

public int write (ByteBuffer audioData, 
                int sizeInBytes, 
                int writeMode)

Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). The audioData in ByteBuffer should match the format specified in the AudioTrack constructor.

In static buffer mode, copies the data to the buffer starting at offset 0, and the write mode is ignored. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`ByteBuffer`: the buffer that holds the data to write, starting at the position reported by `audioData.position()`. Note that upon return, the buffer position (`audioData.position()`) will have been advanced to reflect the amount of data that was successfully written to the AudioTrack. This value cannot be `null`.
`sizeInBytes`	`int`: number of bytes to write. It is recommended but not enforced that the number of bytes requested be a multiple of the frame size (sample size in bytes multiplied by the channel count). Note this may differ from `audioData.remaining()`, but cannot exceed it.
`writeMode`	`int`: one of `WRITE_BLOCKING`, `WRITE_NON_BLOCKING`. It has no effect in static mode. With `WRITE_BLOCKING`, the write will block until all data has been written to the audio sink. With `WRITE_NON_BLOCKING`, the write will return immediately after queuing as much audio data for playback as possible without blocking. Value is `WRITE_BLOCKING`, or `WRITE_NON_BLOCKING`

Returns

int

zero or the positive number of bytes that were written, or one of the following error codes.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

write

Added in API level 23

public int write (ByteBuffer audioData, 
                int sizeInBytes, 
                int writeMode, 
                long timestamp)

Writes the audio data to the audio sink for playback in streaming mode on a HW_AV_SYNC track. The blocking behavior will depend on the write mode.

Parameters
`audioData`	`ByteBuffer`: the buffer that holds the data to write, starting at the position reported by `audioData.position()`. Note that upon return, the buffer position (`audioData.position()`) will have been advanced to reflect the amount of data that was successfully written to the AudioTrack. This value cannot be `null`.
`sizeInBytes`	`int`: number of bytes to write. It is recommended but not enforced that the number of bytes requested be a multiple of the frame size (sample size in bytes multiplied by the channel count). Note this may differ from `audioData.remaining()`, but cannot exceed it.
`writeMode`	`int`: one of `WRITE_BLOCKING`, `WRITE_NON_BLOCKING`. With `WRITE_BLOCKING`, the write will block until all data has been written to the audio sink. With `WRITE_NON_BLOCKING`, the write will return immediately after queuing as much audio data for playback as possible without blocking. Value is `WRITE_BLOCKING`, or `WRITE_NON_BLOCKING`
`timestamp`	`long`: The timestamp, in nanoseconds, of the first decodable audio frame in the provided audioData.

Returns

int

zero or the positive number of bytes that were written, or one of the following error codes.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

write

Added in API level 23

public int write (short[] audioData, 
                int offsetInShorts, 
                int sizeInShorts, 
                int writeMode)

In static buffer mode, copies the data to the buffer starting at offset 0. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`short`: the array that holds the data to write. This value cannot be `null`.
`offsetInShorts`	`int`: the offset expressed in shorts in audioData where the data to write starts. Must not be negative, or cause the data access to go out of bounds of the array.
`sizeInShorts`	`int`: the number of shorts to read in audioData after the offset. Must not be negative, or cause the data access to go out of bounds of the array.
`writeMode`	`int`: one of `WRITE_BLOCKING`, `WRITE_NON_BLOCKING`. It has no effect in static mode. With `WRITE_BLOCKING`, the write will block until all data has been written to the audio sink. With `WRITE_NON_BLOCKING`, the write will return immediately after queuing as much audio data for playback as possible without blocking. Value is `WRITE_BLOCKING`, or `WRITE_NON_BLOCKING`

Returns

int

zero or the positive number of shorts that were written, or one of the following error codes. The number of shorts will be a multiple of the channel count not to exceed sizeInShorts.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

write

Added in API level 23

public int write (byte[] audioData, 
                int offsetInBytes, 
                int sizeInBytes, 
                int writeMode)

In static buffer mode, copies the data to the buffer starting at offset 0, and the write mode is ignored. Note that the actual playback of this data might occur after this function returns.

Parameters
`audioData`	`byte`: the array that holds the data to play. This value cannot be `null`.
`offsetInBytes`	`int`: the offset expressed in bytes in audioData where the data to write starts. Must not be negative, or cause the data access to go out of bounds of the array.
`sizeInBytes`	`int`: the number of bytes to write in audioData after the offset. Must not be negative, or cause the data access to go out of bounds of the array.
`writeMode`	`int`: one of `WRITE_BLOCKING`, `WRITE_NON_BLOCKING`. It has no effect in static mode. With `WRITE_BLOCKING`, the write will block until all data has been written to the audio sink. With `WRITE_NON_BLOCKING`, the write will return immediately after queuing as much audio data for playback as possible without blocking. Value is `WRITE_BLOCKING`, or `WRITE_NON_BLOCKING`

Returns

int

zero or the positive number of bytes that were written, or one of the following error codes. The number of bytes will be a multiple of the frame size in bytes not to exceed sizeInBytes.

ERROR_INVALID_OPERATION if the track isn't properly initialized
ERROR_BAD_VALUE if the parameters don't resolve to valid data and indexes
ERROR_DEAD_OBJECT if the AudioTrack is not valid anymore and needs to be recreated. The dead object error code is not returned if some data was successfully transferred. In this case, the error is returned at the next write()
ERROR in case of other error

Protected methods

finalize

Added in API level 3

protected void finalize ()

Called by the garbage collector on an object when garbage collection determines that there are no more references to the object. A subclass overrides the finalize method to dispose of system resources or to perform other cleanup.

The general contract of finalize is that it is invoked if and when the Java virtual machine has determined that there is no longer any means by which this object can be accessed by any thread that has not yet died, except as a result of an action taken by the finalization of some other object or class which is ready to be finalized. The finalize method may take any action, including making this object available again to other threads; the usual purpose of finalize, however, is to perform cleanup actions before the object is irrevocably discarded. For example, the finalize method for an object that represents an input/output connection might perform explicit I/O transactions to break the connection before the object is permanently discarded.

The finalize method of class Object performs no special action; it simply returns normally. Subclasses of Object may override this definition.

The Java programming language does not guarantee which thread will invoke the finalize method for any given object. It is guaranteed, however, that the thread that invokes finalize will not be holding any user-visible synchronization locks when finalize is invoked. If an uncaught exception is thrown by the finalize method, the exception is ignored and finalization of that object terminates.

After the finalize method has been invoked for an object, no further action is taken until the Java virtual machine has again determined that there is no longer any means by which this object can be accessed by any thread that has not yet died, including possible actions by other objects or classes which are ready to be finalized, at which point the object may be discarded.

The finalize method is never invoked more than once by a Java virtual machine for any given object.

Any exception thrown by the finalize method causes the finalization of this object to be halted, but is otherwise ignored.

getNativeFrameCount

Added in API level 3
Deprecated in API level 19

protected int getNativeFrameCount ()

This method was deprecated in API level 19.
Use the identical public method getBufferSizeInFrames() instead.

Returns the frame count of the native AudioTrack buffer.

Returns
`int`	current size in frames of the `AudioTrack` buffer.

Throws
	java.lang.IllegalStateException

setState

Added in API level 3
Deprecated in API level 19

protected void setState (int state)

This method was deprecated in API level 19.
Only accessible by subclasses, which are not recommended for AudioTrack.

Sets the initialization state of the instance. This method was originally intended to be used in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. However, subclasses of AudioTrack are no longer recommended, so this method is obsolete.

Parameters
`state`	`int`: the state of the AudioTrack instance

AudioTrack

Summary

Nested classes

Constants

Public constructors

Public methods

Protected methods

Inherited methods

Constants

DUAL_MONO_MODE_LL

DUAL_MONO_MODE_LR

DUAL_MONO_MODE_OFF

DUAL_MONO_MODE_RR

ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR

ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER

ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT

ENCAPSULATION_MODE_ELEMENTARY_STREAM

ENCAPSULATION_MODE_NONE

ERROR

ERROR_BAD_VALUE

ERROR_DEAD_OBJECT

ERROR_INVALID_OPERATION

MODE_STATIC

MODE_STREAM

PERFORMANCE_MODE_LOW_LATENCY

PERFORMANCE_MODE_NONE

PERFORMANCE_MODE_POWER_SAVING

PLAYSTATE_PAUSED

PLAYSTATE_PLAYING

PLAYSTATE_STOPPED

STATE_INITIALIZED

STATE_NO_STATIC_DATA

STATE_UNINITIALIZED

SUCCESS

SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT

SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL

SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT

WRITE_BLOCKING

WRITE_NON_BLOCKING

Public constructors

AudioTrack

AudioTrack

AudioTrack

Public methods

addOnCodecFormatChangedListener

addOnRoutingChangedListener

addOnRoutingChangedListener

attachAuxEffect

createVolumeShaper

flush

getAudioAttributes

getAudioDescriptionMixLeveldB

getAudioFormat

getAudioSessionId

getBufferCapacityInFrames

getBufferSizeInFrames

getChannelConfiguration

getChannelCount

getDualMonoMode

getFormat

getLogSessionId

getMaxVolume

getMetrics

getMinBufferSize

getMinVolume

getNativeOutputSampleRate

getNotificationMarkerPosition

getOffloadDelay

getOffloadPadding

getPerformanceMode

getPlayState

getPlaybackHeadPosition

getPlaybackParams

getPlaybackRate

getPositionNotificationPeriod

getPreferredDevice

getRoutedDevice

getSampleRate

getStartThresholdInFrames

getState