AudioTrack

Denotes a generic operation failure.

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

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

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

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

indicates AudioTrack state is paused

indicates AudioTrack state is playing

indicates AudioTrack state is stopped

State of an AudioTrack that is ready to be used.

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

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

Denotes a successful operation.

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

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).

Class constructor.

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.

Class constructor with AudioAttributes and AudioFormat.

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.

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.

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

Returns the audio session ID.

Returns the configured channel configuration. See CHANNEL_OUT_MONO and CHANNEL_OUT_STEREO.

Returns the configured number of channels.

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 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.

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 output sample rate in Hz for the specified stream type.

Returns marker position expressed in frames.

Returns the playback state of the AudioTrack instance.

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().

Returns the current playback rate in Hz.

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

Returns the configured audio data sample rate in Hz

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 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.

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.

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.

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

Releases the native AudioTrack resources.

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.

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)

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.

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

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.

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.

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.

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).

Sets the period for the periodic notification event.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.