Android APIs
public class

AudioTrack

extends Object
java.lang.Object
   ↳ android.media.AudioTrack

Class Overview

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
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. 
Constants
int ERROR Denotes a generic operation failure.
int ERROR_BAD_VALUE Denotes a failure due to the use of an invalid value.
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 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 WRITE_BLOCKING The write mode indicating the write operation will block until all data has been written, to be used in write(ByteBuffer, int, int)
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 in write(ByteBuffer, int, int).
Public Constructors
AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, int bufferSizeInBytes, int mode)
Class constructor.
AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, int bufferSizeInBytes, int mode, int sessionId)
Class constructor with audio session.
Public Methods
int attachAuxEffect(int effectId)
Attaches an auxiliary effect to the audio track.
void flush()
Flushes the audio data currently queued for playback.
int getAudioFormat()
Returns the configured audio data format.
int getAudioSessionId()
Returns the audio session ID.
int getChannelConfiguration()
Returns the configured channel configuration.
int getChannelCount()
Returns the configured number of channels.
static float getMaxVolume()
Returns the maximum gain value, which is greater than or equal to 1.0.
static int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat)
Returns the minimum buffer size required for the successful creation of 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 getPlayState()
Returns the playback state of the AudioTrack instance.
int getPlaybackHeadPosition()
Returns the playback head position expressed in frames.
int getPlaybackRate()
Returns the current playback rate in Hz.
int getPositionNotificationPeriod()
Returns the notification update period expressed in frames.
int getSampleRate()
Returns the configured audio data sample rate in Hz
int getState()
Returns the state of the AudioTrack instance.
int getStreamType()
Returns the type of audio stream this AudioTrack is configured for.
boolean getTimestamp(AudioTimestamp timestamp)
Poll for a timestamp on demand.
void pause()
Pauses the playback of the audio data.
void play()
Starts playing an AudioTrack.
void release()
Releases the native AudioTrack resources.
int reloadStaticData()
Notifies the native resource to reuse the audio data already loaded in the native layer, that is to rewind to start of buffer.
int setAuxEffectSendLevel(float level)
Sets the send level of the audio track to the attached auxiliary effect attachAuxEffect(int).
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.
int setPlaybackHeadPosition(int positionInFrames)
Sets the playback head position.
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.
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.
int setPlaybackRate(int sampleRateInHz)
Sets the playback sample rate for this track.
int setPositionNotificationPeriod(int periodInFrames)
Sets the period for the periodic notification event.
int setStereoVolume(float leftGain, float rightGain)
This method is deprecated. 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.
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(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(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).
Protected Methods
void finalize()
Invoked when the garbage collector has detected that this instance is no longer reachable.
int getNativeFrameCount()
This method was deprecated in API level 19. Only accessible by subclasses, which are not recommended for AudioTrack. See getProperty(String) for key PROPERTY_OUTPUT_FRAMES_PER_BUFFER.
void setState(int state)
This method was deprecated in API level 19. Only accessible by subclasses, which are not recommended for AudioTrack.
[Expand]
Inherited Methods
From class java.lang.Object

Constants

public static final int ERROR

Added in API level 3

Denotes a generic operation failure.

Constant Value: -1 (0xffffffff)

public static final int ERROR_BAD_VALUE

Added in API level 3

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

Constant Value: -2 (0xfffffffe)

public static final int ERROR_INVALID_OPERATION

Added in API level 3

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

Constant Value: -3 (0xfffffffd)

public static final int MODE_STATIC

Added in API level 3

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

Constant Value: 0 (0x00000000)

public static final int MODE_STREAM

Added in API level 3

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

Constant Value: 1 (0x00000001)

public static final int PLAYSTATE_PAUSED

Added in API level 3

indicates AudioTrack state is paused

Constant Value: 2 (0x00000002)

public static final int PLAYSTATE_PLAYING

Added in API level 3

indicates AudioTrack state is playing

Constant Value: 3 (0x00000003)

public static final int PLAYSTATE_STOPPED

Added in API level 3

indicates AudioTrack state is stopped

Constant Value: 1 (0x00000001)

public static final int STATE_INITIALIZED

Added in API level 3

State of an AudioTrack that is ready to be used.

Constant Value: 1 (0x00000001)

public static final int STATE_NO_STATIC_DATA

Added in API level 3

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

Constant Value: 2 (0x00000002)

public static final int STATE_UNINITIALIZED

Added in API level 3

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

Constant Value: 0 (0x00000000)

public static final int SUCCESS

Added in API level 3

Denotes a successful operation.

Constant Value: 0 (0x00000000)

public static final int WRITE_BLOCKING

The write mode indicating the write operation will block until all data has been written, to be used in write(ByteBuffer, int, int)

Constant Value: 0 (0x00000000)

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 in write(ByteBuffer, int, int).

Constant Value: 1 (0x00000001)

Public Constructors

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

Added in API level 3

Class constructor.

Parameters
streamType the type of the audio stream. See STREAM_VOICE_CALL, STREAM_SYSTEM, STREAM_RING, STREAM_MUSIC, STREAM_ALARM, and STREAM_NOTIFICATION.
sampleRateInHz the initial source sample rate expressed in Hz.
channelConfig describes the configuration of the audio channels. See CHANNEL_OUT_MONO and CHANNEL_OUT_STEREO
audioFormat the format in which the audio data is represented. See ENCODING_PCM_16BIT, ENCODING_PCM_8BIT, and ENCODING_PCM_FLOAT.
bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is read from for playback. If track's creation mode is MODE_STREAM, you can write data into this buffer in chunks less than or equal to this size, and it is typical to use chunks of 1/2 of the total size to permit double-buffering. 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. See getMinBufferSize(int, int, int) to determine the minimum required buffer size for the successful creation of an AudioTrack instance in streaming mode. Using values smaller than getMinBufferSize() will result in an initialization failure.
mode streaming or static buffer. See MODE_STATIC and MODE_STREAM

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

Added in API level 9

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 the type of the audio stream. See STREAM_VOICE_CALL, STREAM_SYSTEM, STREAM_RING, STREAM_MUSIC, STREAM_ALARM, and STREAM_NOTIFICATION.
sampleRateInHz the initial source sample rate expressed in Hz.
channelConfig describes the configuration of the audio channels. See CHANNEL_OUT_MONO and CHANNEL_OUT_STEREO
audioFormat the format in which the audio data is represented. See ENCODING_PCM_16BIT and ENCODING_PCM_8BIT, and ENCODING_PCM_FLOAT.
bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read from for playback. If using the AudioTrack in streaming mode, you can write data into this buffer in smaller chunks than this size. If using the AudioTrack in static mode, this is the maximum size of the sound that will be played for this instance. See getMinBufferSize(int, int, int) to determine the minimum required buffer size for the successful creation of an AudioTrack instance in streaming mode. Using values smaller than getMinBufferSize() will result in an initialization failure.
mode streaming or static buffer. See MODE_STATIC and MODE_STREAM
sessionId Id of audio session the AudioTrack must be attached to

Public Methods

public int attachAuxEffect (int effectId)

Added in API level 9

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 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 system wide unique id of the effect to attach
Returns

public void flush ()

Added in API level 3

Flushes the audio data currently queued for playback. Any data that has not been played back will be discarded. No-op if not stopped or paused, or if the track's creation mode is not MODE_STREAM.

public int getAudioFormat ()

Added in API level 3

Returns the configured audio data format. See ENCODING_PCM_16BIT and ENCODING_PCM_8BIT.

public int getAudioSessionId ()

Added in API level 9

Returns the audio session ID.

Returns
  • the ID of the audio session this AudioTrack belongs to.

public int getChannelConfiguration ()

Added in API level 3

Returns the configured channel configuration. See CHANNEL_OUT_MONO and CHANNEL_OUT_STEREO.

public int getChannelCount ()

Added in API level 3

Returns the configured number of channels.

public static float getMaxVolume ()

Added in API level 3

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
  • the maximum value, which is greater than or equal to 1.0.

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

Added in API level 3

Returns the minimum buffer size required for the successful creation of an AudioTrack object to be created in the MODE_STREAM mode. 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 the source sample rate expressed in Hz.
channelConfig describes the configuration of the audio channels. See CHANNEL_OUT_MONO and CHANNEL_OUT_STEREO
audioFormat the format in which the audio data is represented. See ENCODING_PCM_16BIT and ENCODING_PCM_8BIT, and ENCODING_PCM_FLOAT.
Returns
  • 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.

public static float getMinVolume ()

Added in API level 3

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
  • the minimum value, which is the constant 0.0.

public static int getNativeOutputSampleRate (int streamType)

Added in API level 3

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

public int getNotificationMarkerPosition ()

Added in API level 3

Returns marker position expressed in frames.

Returns

public int getPlayState ()

Added in API level 3

Returns the playback state of the AudioTrack instance.

public int getPlaybackHeadPosition ()

Added in API level 3

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(), reload(), and stop().

public int getPlaybackRate ()

Added in API level 3

Returns the current playback rate in Hz.

public int getPositionNotificationPeriod ()

Added in API level 3

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

public int getSampleRate ()

Added in API level 3

Returns the configured audio data sample rate in Hz

public int getState ()

Added in API level 3

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.

public int getStreamType ()

Added in API level 3

Returns the type of audio stream this AudioTrack is configured for. Compare the result against STREAM_VOICE_CALL, STREAM_SYSTEM, STREAM_RING, STREAM_MUSIC, STREAM_ALARM, STREAM_NOTIFICATION, or STREAM_DTMF.

public boolean getTimestamp (AudioTimestamp timestamp)

Added in API level 19

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 once per second until the reported timestamps show that the audio clock is stable. Thereafter, 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 a reference to a non-null AudioTimestamp instance allocated and owned by caller.
Returns
  • true if a timestamp is available, or false if no timestamp is available. If a timestamp if 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.

public void pause ()

Added in API level 3

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.

public void play ()

Added in API level 3

Starts playing an AudioTrack. If track's creation mode is MODE_STATIC, you must have called write() prior.

public void release ()

Added in API level 3

Releases the native AudioTrack resources.

public int reloadStaticData ()

Added in API level 3

Notifies the native resource to reuse the audio data already loaded in the native layer, that is to rewind to start of buffer. The track's creation mode must be MODE_STATIC.

Returns

public int setAuxEffectSendLevel (float level)

Added in API level 9

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 linear send level
Returns

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

Added in API level 3

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 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.
endInFrames 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.
loopCount the number of times the loop is looped. A value of -1 means infinite looping, and 0 disables looping.
Returns

public int setNotificationMarkerPosition (int markerInFrames)

Added in API level 3

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

Parameters
markerInFrames 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

public int setPlaybackHeadPosition (int positionInFrames)

Added in API level 3

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

Parameters
positionInFrames playback head position expressed in frames Zero corresponds to start of buffer. The position must not be greater than the buffer size in frames, or negative.
Returns

public void setPlaybackPositionUpdateListener (AudioTrack.OnPlaybackPositionUpdateListener listener)

Added in API level 3

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.

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

Added in API level 3

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 the Handler that will receive the event notification messages.

public int setPlaybackRate (int sampleRateInHz)

Added in API level 3

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).

Parameters
sampleRateInHz the sample rate expressed in Hz
Returns

public int setPositionNotificationPeriod (int periodInFrames)

Added in API level 3

Sets the period for the periodic notification event.

Parameters
periodInFrames update period expressed in frames
Returns

public int setStereoVolume (float leftGain, float rightGain)

Added in API level 3

This method is deprecated.
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 output gain for the left channel.
rightGain output gain for the right channel
Returns

public int setVolume (float gain)

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

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.

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 output gain for all channels.
Returns

public void stop ()

Added in API level 3

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.

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). In static buffer mode, copies the data to the buffer starting at offset 0, and the write mode is ignored. In streaming mode, the blocking behavior will depend on the write mode.

Note that the actual playback of this data might occur after this function returns. This function is thread safe with respect to stop() calls, in which case all of the specified data might not be written to the audio sink.

Parameters
audioData the array that holds the data to play. 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.
offsetInFloats the offset, expressed as a number of floats, in audioData where the data to play starts.
sizeInFloats the number of floats to read in audioData after the offset.
writeMode 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.
Returns

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). In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write mode is ignored. In streaming mode, the blocking behavior will depend on the write mode.

Parameters
audioData the buffer that holds the data to play, 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.
sizeInBytes number of bytes to write.
Note this may differ from audioData.remaining(), but cannot exceed it.
writeMode 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.
Returns

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

Added in API level 3

Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). In streaming mode, will block until all data has been written to the audio sink. 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. This function is thread safe with respect to stop() calls, in which case all of the specified data might not be written to the audio sink.

Parameters
audioData the array that holds the data to play.
offsetInShorts the offset expressed in shorts in audioData where the data to play starts.
sizeInShorts the number of shorts to read in audioData after the offset.
Returns

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

Added in API level 3

Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). In streaming mode, will block until all data has been written to the audio sink. 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. This function is thread safe with respect to stop() calls, in which case all of the specified data might not be written to the audio sink.

Parameters
audioData the array that holds the data to play.
offsetInBytes the offset expressed in bytes in audioData where the data to play starts.
sizeInBytes the number of bytes to read in audioData after the offset.
Returns

Protected Methods

protected void finalize ()

Added in API level 3

Invoked when the garbage collector has detected that this instance is no longer reachable. The default implementation does nothing, but this method can be overridden to free resources.

Note that objects that override finalize are significantly more expensive than objects that don't. Finalizers may be run a long time after the object is no longer reachable, depending on memory pressure, so it's a bad idea to rely on them for cleanup. Note also that finalizers are run on a single VM-wide finalizer thread, so doing blocking work in a finalizer is a bad idea. A finalizer is usually only necessary for a class that has a native peer and needs to call a native method to destroy that peer. Even then, it's better to provide an explicit close method (and implement Closeable), and insist that callers manually dispose of instances. This works well for something like files, but less well for something like a BigInteger where typical calling code would have to deal with lots of temporaries. Unfortunately, code that creates lots of temporaries is the worst kind of code from the point of view of the single finalizer thread.

If you must use finalizers, consider at least providing your own ReferenceQueue and having your own thread process that queue.

Unlike constructors, finalizers are not automatically chained. You are responsible for calling super.finalize() yourself.

Uncaught exceptions thrown by finalizers are ignored and do not terminate the finalizer thread. See Effective Java Item 7, "Avoid finalizers" for more.

protected int getNativeFrameCount ()

Added in API level 3

This method was deprecated in API level 19.
Only accessible by subclasses, which are not recommended for AudioTrack. See getProperty(String) for key PROPERTY_OUTPUT_FRAMES_PER_BUFFER.

Returns the "native frame count", derived from the bufferSizeInBytes specified at creation time and converted to frame units. If track's creation mode is MODE_STATIC, it is equal to the specified bufferSizeInBytes converted to frame units. If track's creation mode is MODE_STREAM, it is typically greater than or equal to the specified bufferSizeInBytes converted to frame units; it may be rounded up to a larger value if needed by the target device implementation.

protected void setState (int state)

Added in API level 3

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 the state of the AudioTrack instance