AudioTrack
open class AudioTrack : AudioRouting, VolumeAutomation
kotlin.Any | |
↳ | 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.
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 | |
---|---|
open |
Builder class for |
abstract |
Interface definition for a listener for codec format changes. |
abstract |
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. |
abstract |
Defines the interface by which applications can receive notifications of routing changes for the associated |
abstract |
Abstract class to receive event notifications about the stream playback in offloaded mode. |
Constants | |
---|---|
static Int |
This mode indicates that a stereo stream should be presented with the left audio channel replicated into the right audio channel. |
static Int |
This mode indicates that a stereo stream should be presented with the left and right audio channels blended together and delivered to both channels. |
static Int |
This mode disables any Dual Mono presentation effect. |
static Int |
This mode indicates that a stereo stream should be presented with the right audio channel replicated into the left audio channel. |
static Int |
Encapsulation metadata type for DVB AD descriptor. |
static Int |
Encapsulation metadata type for framework tuner information. |
static Int |
Encapsulation metadata type for placement of supplementary audio. |
static Int |
This mode indicates metadata encapsulation with an elementary stream payload. |
static Int |
This mode indicates no metadata encapsulation, which is the default mode for sending audio data through |
static Int |
Denotes a generic operation failure. |
static Int |
Denotes a failure due to the use of an invalid value. |
static Int |
An error code indicating that the object reporting it is no longer valid and needs to be recreated. |
static Int |
Denotes a failure due to the improper use of a method. |
static Int |
Creation mode where audio data is transferred from Java to the native layer only once before the audio starts playing. |
static Int |
Creation mode where audio data is streamed from Java to the native layer as the audio is playing. |
static Int |
Low latency performance mode for an |
static Int |
Default performance mode for an |
static Int |
Power saving performance mode for an |
static Int |
indicates AudioTrack state is paused |
static Int |
indicates AudioTrack state is playing |
static Int |
indicates AudioTrack state is stopped |
static Int |
State of an AudioTrack that is ready to be used. |
static Int |
State of a successfully initialized AudioTrack that uses static data, but that hasn't received that data yet. |
static Int |
State of an AudioTrack that was not successfully initialized upon creation. |
static Int |
Denotes a successful operation. |
static Int |
Supplementary audio placement left. |
static Int |
Supplementary audio placement normal. |
static Int |
Supplementary audio placement right. |
static Int |
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 |
static 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 as the actual value of the writeMode parameter in |
Public constructors | |
---|---|
AudioTrack(streamType: Int, sampleRateInHz: Int, channelConfig: Int, audioFormat: Int, bufferSizeInBytes: Int, mode: Int) Class constructor. |
|
AudioTrack(streamType: Int, sampleRateInHz: Int, channelConfig: Int, audioFormat: Int, bufferSizeInBytes: Int, mode: Int, sessionId: Int) Class constructor with audio session. |
|
AudioTrack(attributes: AudioAttributes!, format: AudioFormat!, bufferSizeInBytes: Int, mode: Int, sessionId: Int) Class constructor with |
Public methods | |
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open Unit |
addOnCodecFormatChangedListener(executor: Executor, listener: AudioTrack.OnCodecFormatChangedListener) Adds an |
open Unit |
addOnRoutingChangedListener(listener: AudioRouting.OnRoutingChangedListener!, handler: Handler!) Adds an |
open Unit |
addOnRoutingChangedListener(listener: AudioTrack.OnRoutingChangedListener!, handler: Handler!) Adds an |
open Int |
attachAuxEffect(effectId: Int) Attaches an auxiliary effect to the audio track. |
open VolumeShaper |
createVolumeShaper(configuration: VolumeShaper.Configuration) Returns a |
open Unit |
flush() Flushes the audio data currently queued for playback. |
open AudioAttributes |
Returns the |
open Float |
Returns the Audio Description mix level in dB. |
open Int |
Returns the configured audio data encoding. |
open Int |
Returns the audio session ID. |
open Int |
Returns the maximum size of the |
open Int |
Returns the effective size of the |
open Int |
Returns the configured channel position mask. |
open Int |
Returns the configured number of channels. |
open Int |
Returns the Dual Mono mode presentation setting. |
open AudioFormat |
Returns the configured |
open LogSessionId |
Returns the |
open static Float |
Returns the maximum gain value, which is greater than or equal to 1. |
open PersistableBundle! |
Return Metrics data about the current AudioTrack instance. |
open static Int |
getMinBufferSize(sampleRateInHz: Int, channelConfig: Int, audioFormat: Int) Returns the estimated minimum buffer size required for an AudioTrack object to be created in the |
open static Float |
Returns the minimum gain value, which is the constant 0. |
open static Int |
getNativeOutputSampleRate(streamType: Int) Returns the output sample rate in Hz for the specified stream type. |
open Int |
Returns marker position expressed in frames. |
open Int |
Return the decoder delay of an offloaded track, expressed in frames, previously set with |
open Int |
Return the decoder padding of an offloaded track, expressed in frames, previously set with |
open Int |
Returns the current performance mode of the |
open Int |
Returns the playback state of the AudioTrack instance. |
open Int |
Returns the playback head position expressed in frames. |
open PlaybackParams |
Returns the current playback parameters. |
open Int |
Returns the current playback sample rate rate in Hz. |
open Int |
Returns the notification update period expressed in frames. |
open AudioDeviceInfo! |
Returns the selected output specified by |
open AudioDeviceInfo! |
Returns an |
open Int |
Returns the configured audio source sample rate in Hz. |
open Int |
Returns the streaming start threshold of the |
open Int |
getState() Returns the state of the AudioTrack instance. |
open Int |
Returns the volume stream type of this AudioTrack. |
open Boolean |
getTimestamp(timestamp: AudioTimestamp!) Poll for a timestamp on demand. |
open Int |
Returns the number of underrun occurrences in the application-level write buffer since the AudioTrack was created. |
open static Boolean |
isDirectPlaybackSupported(format: AudioFormat, attributes: AudioAttributes) Returns whether direct playback of an audio format with the provided attributes is currently supported on the system. |
open Boolean |
Returns whether the track was built with |
open Unit |
pause() Pauses the playback of the audio data. |
open Unit |
play() Starts playing an AudioTrack. |
open Unit |
registerStreamEventCallback(executor: Executor, eventCallback: AudioTrack.StreamEventCallback) Registers a callback for the notification of stream events. |
open Unit |
release() Releases the native AudioTrack resources. |
open Int |
Sets the playback head position within the static buffer to zero, that is it rewinds to start of static buffer. |
open Unit |
Removes an |
open Unit |
Removes an |
open Unit |
Removes an |
open Boolean |
setAudioDescriptionMixLeveldB(level: Float) Sets the Audio Description mix level in dB. |
open Int |
setAuxEffectSendLevel(level: Float) Sets the send level of the audio track to the attached auxiliary effect |
open Int |
setBufferSizeInFrames(bufferSizeInFrames: Int) Limits the effective size of the |
open Boolean |
setDualMonoMode(dualMonoMode: Int) Sets the Dual Mono mode presentation on the output device. |
open Unit |
setLogSessionId(logSessionId: LogSessionId) Sets a |
open Int |
setLoopPoints(startInFrames: Int, endInFrames: Int, loopCount: Int) Sets the loop points and the loop count. |
open Int |
setNotificationMarkerPosition(markerInFrames: Int) Sets the position of the notification marker. |
open Unit |
setOffloadDelayPadding(delayInFrames: Int, paddingInFrames: Int) Configures the delay and padding values for the current compressed stream playing in offload mode. |
open Unit |
Declares that the last write() operation on this track provided the last buffer of this stream. |
open Int |
setPlaybackHeadPosition(positionInFrames: Int) Sets the playback head position within the static buffer. |
open Unit |
setPlaybackParams(params: PlaybackParams) Sets the playback parameters. |
open Unit |
Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update. |
open Unit |
setPlaybackPositionUpdateListener(listener: AudioTrack.OnPlaybackPositionUpdateListener!, handler: Handler!) Sets the listener the AudioTrack notifies when a previously set marker is reached or for each periodic playback head position update. |
open Int |
setPlaybackRate(sampleRateInHz: Int) Sets the playback sample rate for this track. |
open Int |
setPositionNotificationPeriod(periodInFrames: Int) Sets the period for the periodic notification event. |
open Boolean |
setPreferredDevice(deviceInfo: AudioDeviceInfo!) Specifies an audio device (via an |
open Int |
setPresentation(presentation: AudioPresentation) Sets the audio presentation. |
open Int |
setStartThresholdInFrames(startThresholdInFrames: Int) Sets the streaming start threshold for an |
open Int |
setStereoVolume(leftGain: Float, rightGain: Float) Sets the specified left and right output gain values on the AudioTrack. |
open Int |
Sets the specified output gain value on all channels of this track. |
open Unit |
stop() Stops playing the audio data. |
open Unit |
unregisterStreamEventCallback(eventCallback: AudioTrack.StreamEventCallback) Unregisters the callback for notification of stream events, previously registered with |
open Int |
Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
write(audioData: ShortArray, offsetInShorts: Int, sizeInShorts: Int) Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
write(audioData: ShortArray, offsetInShorts: Int, sizeInShorts: Int, writeMode: Int) Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
write(audioData: FloatArray, offsetInFloats: Int, sizeInFloats: Int, writeMode: Int) Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
write(audioData: ByteBuffer, sizeInBytes: Int, writeMode: Int) Writes the audio data to the audio sink for playback (streaming mode), or copies audio data for later playback (static buffer mode). |
open Int |
write(audioData: ByteBuffer, sizeInBytes: Int, writeMode: Int, timestamp: Long) Writes the audio data to the audio sink for playback in streaming mode on a HW_AV_SYNC track. |
Protected methods | |
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open Unit |
finalize() |
open Int |
Returns the frame count of the native |
open Unit |
Sets the initialization state of the instance. |
Constants
DUAL_MONO_MODE_LL
static val DUAL_MONO_MODE_LL: Int
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.
Value: 2
DUAL_MONO_MODE_LR
static val DUAL_MONO_MODE_LR: Int
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.
Value: 1
DUAL_MONO_MODE_OFF
static val DUAL_MONO_MODE_OFF: Int
This mode disables any Dual Mono presentation effect.
Value: 0
DUAL_MONO_MODE_RR
static val DUAL_MONO_MODE_RR: Int
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.
Value: 3
ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR
static val ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR: Int
Encapsulation metadata type for DVB AD descriptor. This metadata is formatted per ETSI TS 101 154 Table E.1: AD_descriptor.
Value: 2
ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER
static val ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER: Int
Encapsulation metadata type for framework tuner information. Refer to the Android Media TV Tuner API for details.
Value: 1
ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT
static val ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT: Int
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
.
Value: 3
ENCAPSULATION_MODE_ELEMENTARY_STREAM
static val ENCAPSULATION_MODE_ELEMENTARY_STREAM: Int
This mode indicates metadata encapsulation with an elementary stream payload. Both compressed and PCM format is allowed.
Value: 1
ENCAPSULATION_MODE_NONE
static val ENCAPSULATION_MODE_NONE: Int
This mode indicates no metadata encapsulation, which is the default mode for sending audio data through AudioTrack
.
Value: 0
ERROR_BAD_VALUE
static val ERROR_BAD_VALUE: Int
Denotes a failure due to the use of an invalid value.
Value: -2
ERROR_DEAD_OBJECT
static val ERROR_DEAD_OBJECT: Int
An error code indicating that the object reporting it is no longer valid and needs to be recreated.
Value: -6
ERROR_INVALID_OPERATION
static val ERROR_INVALID_OPERATION: Int
Denotes a failure due to the improper use of a method.
Value: -3
MODE_STATIC
static val MODE_STATIC: Int
Creation mode where audio data is transferred from Java to the native layer only once before the audio starts playing.
Value: 0
MODE_STREAM
static val MODE_STREAM: Int
Creation mode where audio data is streamed from Java to the native layer as the audio is playing.
Value: 1
PERFORMANCE_MODE_LOW_LATENCY
static val PERFORMANCE_MODE_LOW_LATENCY: Int
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.
Value: 1
PERFORMANCE_MODE_NONE
static val PERFORMANCE_MODE_NONE: Int
Default performance mode for an AudioTrack
.
Value: 0
PERFORMANCE_MODE_POWER_SAVING
static val PERFORMANCE_MODE_POWER_SAVING: Int
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.
Value: 2
PLAYSTATE_PAUSED
static val PLAYSTATE_PAUSED: Int
indicates AudioTrack state is paused
Value: 2
PLAYSTATE_PLAYING
static val PLAYSTATE_PLAYING: Int
indicates AudioTrack state is playing
Value: 3
PLAYSTATE_STOPPED
static val PLAYSTATE_STOPPED: Int
indicates AudioTrack state is stopped
Value: 1
STATE_INITIALIZED
static val STATE_INITIALIZED: Int
State of an AudioTrack that is ready to be used.
Value: 1
STATE_NO_STATIC_DATA
static val STATE_NO_STATIC_DATA: Int
State of a successfully initialized AudioTrack that uses static data, but that hasn't received that data yet.
Value: 2
STATE_UNINITIALIZED
static val STATE_UNINITIALIZED: Int
State of an AudioTrack that was not successfully initialized upon creation.
Value: 0
SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT
static val SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT: Int
Supplementary audio placement left.
Value: 1
SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL
static val SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL: Int
Supplementary audio placement normal.
Value: 0
SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT
static val SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT: Int
Supplementary audio placement right.
Value: 2
WRITE_BLOCKING
static val WRITE_BLOCKING: Int
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)
.
Value: 0
WRITE_NON_BLOCKING
static val WRITE_NON_BLOCKING: 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 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)
.
Value: 1
Public constructors
AudioTrack
AudioTrack(
streamType: Int,
sampleRateInHz: Int,
channelConfig: Int,
audioFormat: Int,
bufferSizeInBytes: Int,
mode: Int)
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 If the track's creation mode is |
mode |
Int: streaming or static buffer. See MODE_STATIC and MODE_STREAM |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
AudioTrack
AudioTrack(
streamType: Int,
sampleRateInHz: Int,
channelConfig: Int,
audioFormat: Int,
bufferSizeInBytes: Int,
mode: Int,
sessionId: Int)
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 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 If the track's creation mode is |
mode |
Int: streaming or static buffer. See MODE_STATIC and MODE_STREAM |
sessionId |
Int: Id of audio session the AudioTrack must be attached to |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
AudioTrack
AudioTrack(
attributes: AudioAttributes!,
format: AudioFormat!,
bufferSizeInBytes: Int,
mode: Int,
sessionId: Int)
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 If the track's creation mode is |
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. |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
Public methods
addOnCodecFormatChangedListener
open fun addOnCodecFormatChangedListener(
executor: Executor,
listener: AudioTrack.OnCodecFormatChangedListener
): Unit
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
open fun addOnRoutingChangedListener(
listener: AudioRouting.OnRoutingChangedListener!,
handler: Handler!
): Unit
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. |
addOnRoutingChangedListener
open funaddOnRoutingChangedListener(
listener: AudioTrack.OnRoutingChangedListener!,
handler: Handler!
): Unit
Deprecated: 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. |
attachAuxEffect
open fun attachAuxEffect(effectId: Int): Int
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(). {@see #setAuxEffectSendLevel(float)}.
After creating an auxiliary effect (e.g. android.media.audiofx.EnvironmentalReverb
), retrieve its ID with android.media.audiofx.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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_INVALID_OPERATION , ERROR_BAD_VALUE |
createVolumeShaper
open fun createVolumeShaper(configuration: VolumeShaper.Configuration): VolumeShaper
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 . |
Return | |
---|---|
VolumeShaper |
This value cannot be null . |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
if the configuration is not allowed by the player. |
java.lang.IllegalStateException |
if too many VolumeShaper s are requested or the state of the player does not permit its creation (e.g. player is released). |
flush
open fun flush(): Unit
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
open fun getAudioAttributes(): AudioAttributes
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.
Return | |
---|---|
AudioAttributes |
The AudioAttributes used to configure the AudioTrack. This value cannot be null . |
Exceptions | |
---|---|
java.lang.IllegalStateException |
If the track is not initialized. |
getAudioDescriptionMixLeveldB
open fun getAudioDescriptionMixLeveldB(): Float
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.
Return | |
---|---|
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
open fun getAudioFormat(): Int
Returns the configured audio data encoding. See AudioFormat#ENCODING_PCM_8BIT
, AudioFormat#ENCODING_PCM_16BIT
, and AudioFormat#ENCODING_PCM_FLOAT
.
getAudioSessionId
open fun getAudioSessionId(): Int
Returns the audio session ID.
Return | |
---|---|
Int |
the ID of the audio session this AudioTrack belongs to. |
getBufferCapacityInFrames
open fun getBufferCapacityInFrames(): Int
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.
See also AudioManager#getProperty(String)
for key AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER
.
Return | |
---|---|
Int |
maximum size in frames of the AudioTrack buffer. Value is 0 or greater |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if track is not initialized. |
getBufferSizeInFrames
open fun getBufferSizeInFrames(): Int
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.
See also AudioManager#getProperty(String)
for key AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER
.
Return | |
---|---|
Int |
current size in frames of the AudioTrack buffer. Value is 0 or greater |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if track is not initialized. |
getChannelConfiguration
open fun getChannelConfiguration(): Int
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.
getChannelCount
open fun getChannelCount(): Int
Returns the configured number of channels.
getDualMonoMode
open fun getDualMonoMode(): Int
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.
getFormat
open fun getFormat(): AudioFormat
Returns the configured AudioTrack
format.
Return | |
---|---|
AudioFormat |
an AudioFormat containing the AudioTrack parameters at the time of configuration. This value cannot be null . |
getLogSessionId
open fun getLogSessionId(): LogSessionId
Returns the LogSessionId
.
Return | |
---|---|
LogSessionId |
This value cannot be null . |
getMaxVolume
open static fun getMaxVolume(): Float
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).
Return | |
---|---|
Float |
the maximum value, which is greater than or equal to 1.0. |
getMetrics
open fun getMetrics(): PersistableBundle!
Return Metrics data about the current AudioTrack instance.
Return | |
---|---|
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
open static fun getMinBufferSize(
sampleRateInHz: Int,
channelConfig: Int,
audioFormat: Int
): Int
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 . |
Return | |
---|---|
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
open static fun getMinVolume(): Float
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.
Return | |
---|---|
Float |
the minimum value, which is the constant 0.0. |
getNativeOutputSampleRate
open static fun getNativeOutputSampleRate(streamType: Int): Int
Returns the output sample rate in Hz for the specified stream type.
getNotificationMarkerPosition
open fun getNotificationMarkerPosition(): Int
Returns marker position expressed in frames.
Return | |
---|---|
Int |
marker position in wrapping frame units similar to getPlaybackHeadPosition , or zero if marker is disabled. |
getOffloadDelay
open fun getOffloadDelay(): Int
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)
.
Return | |
---|---|
Int |
decoder delay expressed in frames. Value is 0 or greater |
getOffloadPadding
open fun getOffloadPadding(): Int
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)
.
Return | |
---|---|
Int |
decoder padding expressed in frames. Value is 0 or greater |
getPerformanceMode
open fun getPerformanceMode(): Int
Returns the current performance mode of the AudioTrack
.
Return | |
---|---|
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 android.media.AudioTrack#PERFORMANCE_MODE_NONE , android.media.AudioTrack#PERFORMANCE_MODE_LOW_LATENCY , or android.media.AudioTrack#PERFORMANCE_MODE_POWER_SAVING |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if track is not initialized. |
getPlayState
open fun getPlayState(): Int
Returns the playback state of the AudioTrack instance.
getPlaybackHeadPosition
open fun getPlaybackHeadPosition(): Int
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.
getPlaybackParams
open fun getPlaybackParams(): PlaybackParams
Returns the current playback parameters. See setPlaybackParams(android.media.PlaybackParams)
to set playback parameters
Return | |
---|---|
PlaybackParams |
current PlaybackParams . This value cannot be null . |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if track is not initialized. |
getPlaybackRate
open fun getPlaybackRate(): Int
Returns the current playback sample rate rate in Hz.
getPositionNotificationPeriod
open fun getPositionNotificationPeriod(): Int
Returns the notification update period expressed in frames. Zero means that no position update notifications are being delivered.
getPreferredDevice
open fun getPreferredDevice(): AudioDeviceInfo!
Returns the selected output specified by setPreferredDevice
. Note that this is not guaranteed to correspond to the actual device being used for playback.
getRoutedDevice
open fun getRoutedDevice(): AudioDeviceInfo!
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.
getSampleRate
open fun getSampleRate(): Int
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].
getStartThresholdInFrames
open fun getStartThresholdInFrames(): Int
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.
Return | |
---|---|
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 |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if the track is not initialized or the track is not MODE_STREAM . |
See Also
getState
open fun getState(): Int
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.
getStreamType
open fun getStreamType(): Int
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
.
getTimestamp
open fun getTimestamp(timestamp: AudioTimestamp!): Boolean
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. |
Return | |
---|---|
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
open fun getUnderrunCount(): Int
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.
isDirectPlaybackSupported
open static funisDirectPlaybackSupported(
format: AudioFormat,
attributes: AudioAttributes
): Boolean
Deprecated: 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. |
Return | |
---|---|
Boolean |
true if the given audio format can be played directly. |
isOffloadedPlayback
open fun isOffloadedPlayback(): Boolean
Returns whether the track was built with Builder#setOffloadedPlayback(boolean)
set to true
.
Return | |
---|---|
Boolean |
true if the track is using offloaded playback. |
pause
open fun pause(): Unit
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.
Exceptions | |
---|---|
java.lang.IllegalStateException |
play
open fun play(): Unit
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 android.os.Build.VERSION_CODES#S
the minimum level to start playing can be obtained using getStartThresholdInFrames()
and set with setStartThresholdInFrames(int)
.
Exceptions | |
---|---|
java.lang.IllegalStateException |
if the track isn't properly initialized |
registerStreamEventCallback
open fun registerStreamEventCallback(
executor: Executor,
eventCallback: AudioTrack.StreamEventCallback
): Unit
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 . |
reloadStaticData
open fun reloadStaticData(): Int
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 android.os.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.
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
removeOnCodecFormatChangedListener
open fun removeOnCodecFormatChangedListener(listener: AudioTrack.OnCodecFormatChangedListener): Unit
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
open fun removeOnRoutingChangedListener(listener: AudioRouting.OnRoutingChangedListener!): Unit
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. |
removeOnRoutingChangedListener
open funremoveOnRoutingChangedListener(listener: AudioTrack.OnRoutingChangedListener!): Unit
Deprecated: 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. |
setAudioDescriptionMixLeveldB
open fun setAudioDescriptionMixLeveldB(level: Float): Boolean
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 |
Return | |
---|---|
Boolean |
true on success, false on failure. |
setAuxEffectSendLevel
open fun setAuxEffectSendLevel(level: Float): Int
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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_INVALID_OPERATION , ERROR |
setBufferSizeInFrames
open fun setBufferSizeInFrames(bufferSizeInFrames: Int): Int
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 |
Return | |
---|---|
Int |
the actual buffer size in frames or an error code, ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if track is not initialized. |
setDualMonoMode
open fun setDualMonoMode(dualMonoMode: Int): Boolean
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.
Return | |
---|---|
Boolean |
true on success, false on failure if the output device does not support Dual Mono mode. |
setLogSessionId
open fun setLogSessionId(logSessionId: LogSessionId): Unit
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 . |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if AudioTrack not initialized. |
setLoopPoints
open fun setLoopPoints(
startInFrames: Int,
endInFrames: Int,
loopCount: Int
): Int
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 android.os.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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
setNotificationMarkerPosition
open fun setNotificationMarkerPosition(markerInFrames: Int): Int
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. |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
setOffloadDelayPadding
open fun setOffloadDelayPadding(
delayInFrames: Int,
paddingInFrames: Int
): Unit
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
open fun setOffloadEndOfStream(): Unit
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 {@see Builder#setOffloadedPlayback(boolean)}. Can only be called only if the AudioTrack is in state PLAYSTATE_PLAYING
{@see #getPlayState()}. Use this method in the same thread as any write() operation.
setPlaybackHeadPosition
open fun setPlaybackHeadPosition(positionInFrames: Int): Int
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 android.os.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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
setPlaybackParams
open fun setPlaybackParams(params: PlaybackParams): Unit
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 . |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
if the parameters are invalid or not accepted. |
java.lang.IllegalStateException |
if track is not initialized. |
setPlaybackPositionUpdateListener
open fun setPlaybackPositionUpdateListener(listener: AudioTrack.OnPlaybackPositionUpdateListener!): Unit
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.
Parameters | |
---|---|
listener |
AudioTrack.OnPlaybackPositionUpdateListener!: |
setPlaybackPositionUpdateListener
open fun setPlaybackPositionUpdateListener(
listener: AudioTrack.OnPlaybackPositionUpdateListener!,
handler: Handler!
): Unit
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 | |
---|---|
listener |
AudioTrack.OnPlaybackPositionUpdateListener!: |
handler |
Handler!: the Handler that will receive the event notification messages. |
setPlaybackRate
open fun setPlaybackRate(sampleRateInHz: Int): Int
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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
setPositionNotificationPeriod
open fun setPositionNotificationPeriod(periodInFrames: Int): Int
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. |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_INVALID_OPERATION |
setPreferredDevice
open fun setPreferredDevice(deviceInfo: AudioDeviceInfo!): Boolean
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. |
Return | |
---|---|
Boolean |
true if succesful, false if the specified AudioDeviceInfo is non-null and does not correspond to a valid audio output device. |
setPresentation
open fun setPresentation(presentation: AudioPresentation): Int
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 . |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR , ERROR_BAD_VALUE , ERROR_INVALID_OPERATION |
Exceptions | |
---|---|
java.lang.IllegalArgumentException |
if the audio presentation is null. |
java.lang.IllegalStateException |
if track is not initialized. |
setStartThresholdInFrames
open fun setStartThresholdInFrames(startThresholdInFrames: Int): Int
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 |
Return | |
---|---|
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 |
Exceptions | |
---|---|
java.lang.IllegalStateException |
if the track is not initialized or the track transfer mode is not MODE_STREAM . |
java.lang.IllegalArgumentException |
if startThresholdInFrames is not positive. |
See Also
setStereoVolume
open funsetStereoVolume(
leftGain: Float,
rightGain: Float
): Int
Deprecated: Applications should use setVolume
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 |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_INVALID_OPERATION |
setVolume
open fun setVolume(gain: Float): Int
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
, 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. |
Return | |
---|---|
Int |
error code or success, see SUCCESS , ERROR_INVALID_OPERATION |
stop
open fun stop(): Unit
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.
Exceptions | |
---|---|
java.lang.IllegalStateException |
unregisterStreamEventCallback
open fun unregisterStreamEventCallback(eventCallback: AudioTrack.StreamEventCallback): Unit
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
open fun write(
audioData: ByteArray,
offsetInBytes: Int,
sizeInBytes: Int
): Int
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_8BIT
to correspond to the data in the array. The format can be AudioFormat#ENCODING_PCM_16BIT
, but this is deprecated.
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 |
ByteArray: 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. |
Return | |
---|---|
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.
write(byte[],int,int,int) with writeMode set to WRITE_BLOCKING . |
write
open fun write(
audioData: ByteArray,
offsetInBytes: Int,
sizeInBytes: Int,
writeMode: Int
): Int
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_8BIT
to correspond to the data in the array. The format can be AudioFormat#ENCODING_PCM_16BIT
, but this is deprecated.
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 |
ByteArray: 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 android.media.AudioTrack#WRITE_BLOCKING , or android.media.AudioTrack#WRITE_NON_BLOCKING |
Return | |
---|---|
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.
|
write
open fun write(
audioData: ShortArray,
offsetInShorts: Int,
sizeInShorts: Int
): Int
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_16BIT
to correspond to the data in the array.
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 |
ShortArray: 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. |
Return | |
---|---|
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.
write(short[],int,int,int) with writeMode set to WRITE_BLOCKING . |
write
open fun write(
audioData: ShortArray,
offsetInShorts: Int,
sizeInShorts: Int,
writeMode: Int
): Int
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_16BIT
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. Note that the actual playback of this data might occur after this function returns.
Parameters | |
---|---|
audioData |
ShortArray: 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 android.media.AudioTrack#WRITE_BLOCKING , or android.media.AudioTrack#WRITE_NON_BLOCKING |
Return | |
---|---|
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.
|
write
open fun write(
audioData: FloatArray,
offsetInFloats: Int,
sizeInFloats: Int,
writeMode: Int
): Int
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 |
FloatArray: 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 android.media.AudioTrack#WRITE_BLOCKING , or android.media.AudioTrack#WRITE_NON_BLOCKING |
Return | |
---|---|
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.
|
write
open fun write(
audioData: ByteBuffer,
sizeInBytes: Int,
writeMode: Int
): Int
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 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 |
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 android.media.AudioTrack#WRITE_BLOCKING , or android.media.AudioTrack#WRITE_NON_BLOCKING |
Return | |
---|---|
Int |
zero or the positive number of bytes that were written, or one of the following error codes.
|
write
open fun write(
audioData: ByteBuffer,
sizeInBytes: Int,
writeMode: Int,
timestamp: Long
): Int
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 android.media.AudioTrack#WRITE_BLOCKING , or android.media.AudioTrack#WRITE_NON_BLOCKING |
timestamp |
Long: The timestamp, in nanoseconds, of the first decodable audio frame in the provided audioData. |
Return | |
---|---|
Int |
zero or the positive number of bytes that were written, or one of the following error codes.
|
Protected methods
finalize
protected open fun finalize(): Unit
Exceptions | |
---|---|
java.lang.Throwable |
the Exception raised by this method |
getNativeFrameCount
protected open fungetNativeFrameCount(): Int
Deprecated: Use the identical public method getBufferSizeInFrames()
instead.
Returns the frame count of the native AudioTrack
buffer.
Return | |
---|---|
Int |
current size in frames of the AudioTrack buffer. |
Exceptions | |
---|---|
java.lang.IllegalStateException |
setState
protected open funsetState(state: Int): Unit
Deprecated: 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 |