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 {@code 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:
*
*
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.
*/
public class AudioTrack extends PlayerBase
implements AudioRouting
, VolumeAutomation
{
//---------------------------------------------------------
// Constants
//--------------------
/** Minimum value for a linear gain or auxiliary effect level.
* This value must be exactly equal to 0.0f; do not change it.
*/
private static final float GAIN_MIN = 0.0f;
/** Maximum value for a linear gain or auxiliary effect level.
* This value must be greater than or equal to 1.0f.
*/
private static final float GAIN_MAX = 1.0f;
/** indicates AudioTrack state is stopped */
public static final int PLAYSTATE_STOPPED = 1; // matches SL_PLAYSTATE_STOPPED
/** indicates AudioTrack state is paused */
public static final int PLAYSTATE_PAUSED = 2; // matches SL_PLAYSTATE_PAUSED
/** indicates AudioTrack state is playing */
public static final int PLAYSTATE_PLAYING = 3; // matches SL_PLAYSTATE_PLAYING
/**
* @hide
* indicates AudioTrack state is stopping waiting for NATIVE_EVENT_STREAM_END to
* transition to PLAYSTATE_STOPPED.
* Only valid for offload mode.
*/
private static final int PLAYSTATE_STOPPING = 4;
/**
* @hide
* indicates AudioTrack state is paused from stopping state. Will transition to
* PLAYSTATE_STOPPING if play() is called.
* Only valid for offload mode.
*/
private static final int PLAYSTATE_PAUSED_STOPPING = 5;
// keep these values in sync with android_media_AudioTrack.cpp
/**
* Creation mode where audio data is transferred from Java to the native layer
* only once before the audio starts playing.
*/
public static final int MODE_STATIC = 0;
/**
* Creation mode where audio data is streamed from Java to the native layer
* as the audio is playing.
*/
public static final int MODE_STREAM = 1;
/** @hide */
@IntDef({
MODE_STATIC,
MODE_STREAM
})
@Retention(RetentionPolicy.SOURCE)
public @interface TransferMode {}
/**
* State of an AudioTrack that was not successfully initialized upon creation.
*/
public static final int STATE_UNINITIALIZED = 0;
/**
* State of an AudioTrack that is ready to be used.
*/
public static final int STATE_INITIALIZED = 1;
/**
* State of a successfully initialized AudioTrack that uses static data,
* but that hasn't received that data yet.
*/
public static final int STATE_NO_STATIC_DATA = 2;
/**
* Denotes a successful operation.
*/
public static final int SUCCESS = AudioSystem.SUCCESS;
/**
* Denotes a generic operation failure.
*/
public static final int ERROR = AudioSystem.ERROR;
/**
* Denotes a failure due to the use of an invalid value.
*/
public static final int ERROR_BAD_VALUE = AudioSystem.BAD_VALUE;
/**
* Denotes a failure due to the improper use of a method.
*/
public static final int ERROR_INVALID_OPERATION = AudioSystem.INVALID_OPERATION;
/**
* An error code indicating that the object reporting it is no longer valid and needs to
* be recreated.
*/
public static final int ERROR_DEAD_OBJECT = AudioSystem.DEAD_OBJECT;
/**
* {@link #getTimestampWithStatus(AudioTimestamp)} is called in STOPPED or FLUSHED state,
* or immediately after start/ACTIVE.
* @hide
*/
public static final int ERROR_WOULD_BLOCK = AudioSystem.WOULD_BLOCK;
// Error codes:
// to keep in sync with frameworks/base/core/jni/android_media_AudioTrack.cpp
private static final int ERROR_NATIVESETUP_AUDIOSYSTEM = -16;
private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK = -17;
private static final int ERROR_NATIVESETUP_INVALIDFORMAT = -18;
private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE = -19;
private static final int ERROR_NATIVESETUP_NATIVEINITFAILED = -20;
// Events:
// to keep in sync with frameworks/av/include/media/AudioTrack.h
// Note: To avoid collisions with other event constants,
// do not define an event here that is the same value as
// AudioSystem.NATIVE_EVENT_ROUTING_CHANGE.
/**
* Event id denotes when playback head has reached a previously set marker.
*/
private static final int NATIVE_EVENT_MARKER = 3;
/**
* Event id denotes when previously set update period has elapsed during playback.
*/
private static final int NATIVE_EVENT_NEW_POS = 4;
/**
* Callback for more data
*/
private static final int NATIVE_EVENT_CAN_WRITE_MORE_DATA = 9;
/**
* IAudioTrack tear down for offloaded tracks
* TODO: when received, java AudioTrack must be released
*/
private static final int NATIVE_EVENT_NEW_IAUDIOTRACK = 6;
/**
* Event id denotes when all the buffers queued in AF and HW are played
* back (after stop is called) for an offloaded track.
*/
private static final int NATIVE_EVENT_STREAM_END = 7;
/**
* Event id denotes when the codec format changes.
*
* Note: Similar to a device routing change (AudioSystem.NATIVE_EVENT_ROUTING_CHANGE),
* this event comes from the AudioFlinger Thread / Output Stream management
* (not from buffer indications as above).
*/
private static final int NATIVE_EVENT_CODEC_FORMAT_CHANGE = 100;
private final static String TAG = "android.media.AudioTrack";
/** @hide */
@IntDef({
ENCAPSULATION_MODE_NONE,
ENCAPSULATION_MODE_ELEMENTARY_STREAM,
// ENCAPSULATION_MODE_HANDLE, @SystemApi
})
@Retention(RetentionPolicy.SOURCE)
public @interface EncapsulationMode {}
// Important: The ENCAPSULATION_MODE values must be kept in sync with native header files.
/**
* This mode indicates no metadata encapsulation,
* which is the default mode for sending audio data
* through {@code AudioTrack}.
*/
public static final int ENCAPSULATION_MODE_NONE = 0;
/**
* This mode indicates metadata encapsulation with an elementary stream payload.
* Both compressed and PCM format is allowed.
*/
public static final int ENCAPSULATION_MODE_ELEMENTARY_STREAM = 1;
/**
* This mode indicates metadata encapsulation with a handle payload
* and is set through {@link Builder#setEncapsulationMode(int)}.
* The handle is a 64 bit long, provided by the Tuner API
* in {@link android.os.Build.VERSION_CODES#R}.
* @hide
*/
@SystemApi
@RequiresPermission(android.Manifest.permission.MODIFY_AUDIO_ROUTING)
public static final int ENCAPSULATION_MODE_HANDLE = 2;
/**
* Enumeration of metadata types permitted for use by
* encapsulation mode audio streams.
* @hide
*/
@IntDef(prefix = {"ENCAPSULATION_METADATA_TYPE_"},
value =
{
ENCAPSULATION_METADATA_TYPE_NONE, /* reserved */
ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER,
ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR,
ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT,
})
@Retention(RetentionPolicy.SOURCE)
public @interface EncapsulationMetadataType {}
/**
* Reserved do not use.
* @hide
*/
public static final int ENCAPSULATION_METADATA_TYPE_NONE = 0; // reserved
/**
* Encapsulation metadata type for framework tuner information.
*
* Refer to the Android Media TV Tuner API for details.
*/
public static final int ENCAPSULATION_METADATA_TYPE_FRAMEWORK_TUNER = 1;
/**
* Encapsulation metadata type for DVB AD descriptor.
*
* This metadata is formatted per ETSI TS 101 154 Table E.1: AD_descriptor.
*/
public static final int ENCAPSULATION_METADATA_TYPE_DVB_AD_DESCRIPTOR = 2;
/**
* Encapsulation metadata type for placement of supplementary audio.
*
* A 32 bit integer constant, one of {@link #SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL}, {@link
* #SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT}, {@link #SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT}.
*/
public static final int ENCAPSULATION_METADATA_TYPE_SUPPLEMENTARY_AUDIO_PLACEMENT = 3;
/**
* Enumeration of supplementary audio placement types.
* @hide
*/
@IntDef(prefix = {"SUPPLEMENTARY_AUDIO_PLACEMENT_"},
value =
{
SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL,
SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT,
SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT,
})
@Retention(RetentionPolicy.SOURCE)
public @interface SupplementaryAudioPlacement {}
// Important: The SUPPLEMENTARY_AUDIO_PLACEMENT values must be kept in sync with native header
// files.
/**
* Supplementary audio placement normal.
*/
public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_NORMAL = 0;
/**
* Supplementary audio placement left.
*/
public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_LEFT = 1;
/**
* Supplementary audio placement right.
*/
public static final int SUPPLEMENTARY_AUDIO_PLACEMENT_RIGHT = 2;
/* Dual Mono handling is used when a stereo audio stream
* contains separate audio content on the left and right channels.
* Such information about the content of the stream may be found, for example, in
* ITU T-REC-J.94-201610 A.6.2.3 Component descriptor.
*/
/** @hide */
@IntDef({
DUAL_MONO_MODE_OFF,
DUAL_MONO_MODE_LR,
DUAL_MONO_MODE_LL,
DUAL_MONO_MODE_RR,
})
@Retention(RetentionPolicy.SOURCE)
public @interface DualMonoMode {}
// Important: The DUAL_MONO_MODE values must be kept in sync with native header files.
/**
* This mode disables any Dual Mono presentation effect.
*
*/
public static final int DUAL_MONO_MODE_OFF = 0;
/**
* 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.
*/
public static final int DUAL_MONO_MODE_LR = 1;
/**
* 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.
*/
public static final int DUAL_MONO_MODE_LL = 2;
/**
* 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.
*/
public static final int DUAL_MONO_MODE_RR = 3;
/** @hide */
@IntDef({
WRITE_BLOCKING,
WRITE_NON_BLOCKING
})
@Retention(RetentionPolicy.SOURCE)
public @interface WriteMode {}
/**
* 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
* {@link #write(byte[], int, int, int)}, {@link #write(short[], int, int, int)},
* {@link #write(float[], int, int, int)}, {@link #write(ByteBuffer, int, int)}, and
* {@link #write(ByteBuffer, int, int, long)}.
*/
public final static int WRITE_BLOCKING = 0;
/**
* 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
* {@link #write(ByteBuffer, int, int)}, {@link #write(short[], int, int, int)},
* {@link #write(float[], int, int, int)}, {@link #write(ByteBuffer, int, int)}, and
* {@link #write(ByteBuffer, int, int, long)}.
*/
public final static int WRITE_NON_BLOCKING = 1;
/** @hide */
@IntDef({
PERFORMANCE_MODE_NONE,
PERFORMANCE_MODE_LOW_LATENCY,
PERFORMANCE_MODE_POWER_SAVING
})
@Retention(RetentionPolicy.SOURCE)
public @interface PerformanceMode {}
/**
* Default performance mode for an {@link AudioTrack}.
*/
public static final int PERFORMANCE_MODE_NONE = 0;
/**
* Low latency performance mode for an {@link 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 {@code 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 * {@code write()} call. *
* If this flag is used without specifying a {@code bufferSizeInBytes} then the * {@code AudioTrack}'s actual buffer size may be too small. * It is recommended that a fairly * large buffer should be specified when the {@code AudioTrack} is created. * Then the actual size can be reduced by calling * {@link #setBufferSizeInFrames(int)}. The buffer size can be optimized * by lowering it after each {@code write()} call until the audio glitches, * which is detected by calling * {@link #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. */ public static final int PERFORMANCE_MODE_LOW_LATENCY = 1; /** * Power saving performance mode for an {@link 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 {@code write()} call. * Use {@link #getBufferSizeInFrames()} to determine * the actual buffer size of the {@code AudioTrack} as it may have increased * to accommodate a deeper buffer. */ public static final int PERFORMANCE_MODE_POWER_SAVING = 2; // keep in sync with system/media/audio/include/system/audio-base.h private static final int AUDIO_OUTPUT_FLAG_FAST = 0x4; private static final int AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 0x8; // Size of HW_AV_SYNC track AV header. private static final float HEADER_V2_SIZE_BYTES = 20.0f; //-------------------------------------------------------------------------- // Member variables //-------------------- /** * Indicates the state of the AudioTrack instance. * One of STATE_UNINITIALIZED, STATE_INITIALIZED, or STATE_NO_STATIC_DATA. */ private int mState = STATE_UNINITIALIZED; /** * Indicates the play state of the AudioTrack instance. * One of PLAYSTATE_STOPPED, PLAYSTATE_PAUSED, or PLAYSTATE_PLAYING. */ private int mPlayState = PLAYSTATE_STOPPED; /** * Indicates that we are expecting an end of stream callback following a call * to setOffloadEndOfStream() in a gapless track transition context. The native track * will be restarted automatically. */ private boolean mOffloadEosPending = false; /** * Lock to ensure mPlayState updates reflect the actual state of the object. */ private final Object mPlayStateLock = new Object(); /** * Sizes of the audio buffer. * These values are set during construction and can be stale. * To obtain the current audio buffer frame count use {@link #getBufferSizeInFrames()}. */ private int mNativeBufferSizeInBytes = 0; private int mNativeBufferSizeInFrames = 0; /** * Handler for events coming from the native code. */ private NativePositionEventHandlerDelegate mEventHandlerDelegate; /** * Looper associated with the thread that creates the AudioTrack instance. */ private final Looper mInitializationLooper; /** * The audio data source sampling rate in Hz. * Never {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED}. */ private int mSampleRate; // initialized by all constructors via audioParamCheck() /** * The number of audio output channels (1 is mono, 2 is stereo, etc.). */ private int mChannelCount = 1; /** * The audio channel mask used for calling native AudioTrack */ private int mChannelMask = AudioFormat.CHANNEL_OUT_MONO; /** * The type of the audio stream to play. See * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, * {@link AudioManager#STREAM_ALARM}, {@link AudioManager#STREAM_NOTIFICATION}, and * {@link AudioManager#STREAM_DTMF}. */ @UnsupportedAppUsage private int mStreamType = AudioManager.STREAM_MUSIC; /** * The way audio is consumed by the audio sink, one of MODE_STATIC or MODE_STREAM. */ private int mDataLoadMode = MODE_STREAM; /** * The current channel position mask, as specified on AudioTrack creation. * Can be set simultaneously with channel index mask {@link #mChannelIndexMask}. * May be set to {@link AudioFormat#CHANNEL_INVALID} if a channel index mask is specified. */ private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO; /** * The channel index mask if specified, otherwise 0. */ private int mChannelIndexMask = 0; /** * The encoding of the audio samples. * @see AudioFormat#ENCODING_PCM_8BIT * @see AudioFormat#ENCODING_PCM_16BIT * @see AudioFormat#ENCODING_PCM_FLOAT */ private int mAudioFormat; // initialized by all constructors via audioParamCheck() /** * The AudioAttributes used in configuration. */ private AudioAttributes mConfiguredAudioAttributes; /** * Audio session ID */ private int mSessionId = AUDIO_SESSION_ID_GENERATE; /** * HW_AV_SYNC track AV Sync Header */ private ByteBuffer mAvSyncHeader = null; /** * HW_AV_SYNC track audio data bytes remaining to write after current AV sync header */ private int mAvSyncBytesRemaining = 0; /** * Offset of the first sample of the audio in byte from start of HW_AV_SYNC track AV header. */ private int mOffset = 0; /** * Indicates whether the track is intended to play in offload mode. */ private boolean mOffloaded = false; /** * When offloaded track: delay for decoder in frames */ private int mOffloadDelayFrames = 0; /** * When offloaded track: padding for decoder in frames */ private int mOffloadPaddingFrames = 0; /** * The log session id used for metrics. * {@link LogSessionId#LOG_SESSION_ID_NONE} here means it is not set. */ @NonNull private LogSessionId mLogSessionId = LogSessionId.LOG_SESSION_ID_NONE; private AudioPolicy mAudioPolicy; //-------------------------------- // Used exclusively by native code //-------------------- /** * @hide * Accessed by native methods: provides access to C++ AudioTrack object. */ @SuppressWarnings("unused") @UnsupportedAppUsage protected long mNativeTrackInJavaObj; /** * Accessed by native methods: provides access to the JNI data (i.e. resources used by * the native AudioTrack object, but not stored in it). */ @SuppressWarnings("unused") @UnsupportedAppUsage(maxTargetSdk = Build.VERSION_CODES.R, trackingBug = 170729553) private long mJniData; //-------------------------------------------------------------------------- // Constructor, Finalize //-------------------- /** * Class constructor. * @param streamType the type of the audio stream. See * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. * @param sampleRateInHz the initial source sample rate expressed in Hz. * {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} means to use a route-dependent value * which is usually the sample rate of the sink. * {@link #getSampleRate()} can be used to retrieve the actual sample rate chosen. * @param channelConfig describes the configuration of the audio channels. * See {@link AudioFormat#CHANNEL_OUT_MONO} and * {@link AudioFormat#CHANNEL_OUT_STEREO} * @param audioFormat the format in which the audio data is represented. * See {@link AudioFormat#ENCODING_PCM_16BIT}, * {@link AudioFormat#ENCODING_PCM_8BIT}, * and {@link AudioFormat#ENCODING_PCM_FLOAT}. * @param bufferSizeInBytes 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 {@link #MODE_STATIC}, * this is the maximum length sample, or audio clip, that can be played by this instance. *
If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the AudioTrack to satisfy the application's
* latency requirements.
* If bufferSizeInBytes is less than the
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming AudioTrack to avoid underrun.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
* @throws java.lang.IllegalArgumentException
* @deprecated use {@link Builder} or
* {@link #AudioTrack(AudioAttributes, AudioFormat, int, int, int)} to specify the
* {@link AudioAttributes} instead of the stream type which is only for volume control.
*/
public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode)
throws IllegalArgumentException {
this(streamType, sampleRateInHz, channelConfig, audioFormat,
bufferSizeInBytes, mode, AUDIO_SESSION_ID_GENERATE);
}
/**
* 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 {@link #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.
* @param streamType the type of the audio stream. See
* {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM},
* {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC},
* {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}.
* @param sampleRateInHz the initial source sample rate expressed in Hz.
* {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} means to use a route-dependent value
* which is usually the sample rate of the sink.
* @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_OUT_MONO} and
* {@link AudioFormat#CHANNEL_OUT_STEREO}
* @param audioFormat the format in which the audio data is represented.
* See {@link AudioFormat#ENCODING_PCM_16BIT} and
* {@link AudioFormat#ENCODING_PCM_8BIT},
* and {@link AudioFormat#ENCODING_PCM_FLOAT}.
* @param bufferSizeInBytes 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 {@link #MODE_STATIC}, * this is the maximum length sample, or audio clip, that can be played by this instance. *
If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the AudioTrack to satisfy the application's
* latency requirements.
* If bufferSizeInBytes is less than the
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming AudioTrack to avoid underrun.
* You can write data into this buffer in smaller chunks than this size.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
* @param sessionId Id of audio session the AudioTrack must be attached to
* @throws java.lang.IllegalArgumentException
* @deprecated use {@link Builder} or
* {@link #AudioTrack(AudioAttributes, AudioFormat, int, int, int)} to specify the
* {@link AudioAttributes} instead of the stream type which is only for volume control.
*/
public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode, int sessionId)
throws IllegalArgumentException {
// mState already == STATE_UNINITIALIZED
this((new AudioAttributes.Builder())
.setLegacyStreamType(streamType)
.build(),
(new AudioFormat.Builder())
.setChannelMask(channelConfig)
.setEncoding(audioFormat)
.setSampleRate(sampleRateInHz)
.build(),
bufferSizeInBytes,
mode, sessionId);
deprecateStreamTypeForPlayback(streamType, "AudioTrack", "AudioTrack()");
}
/**
* Class constructor with {@link AudioAttributes} and {@link AudioFormat}.
* @param attributes a non-null {@link AudioAttributes} instance.
* @param format a non-null {@link AudioFormat} instance describing the format of the data
* that will be played through this AudioTrack. See {@link AudioFormat.Builder} for
* configuring the audio format parameters such as encoding, channel mask and sample rate.
* @param bufferSizeInBytes 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 {@link #MODE_STATIC}, * this is the maximum length sample, or audio clip, that can be played by this instance. *
If the track's creation mode is {@link #MODE_STREAM},
* this should be the desired buffer size
* for the Here is an example where
* If the audio attributes are not set with {@link #setAudioAttributes(AudioAttributes)},
* attributes comprising {@link AudioAttributes#USAGE_MEDIA} will be used.
* This is to prevent users of the call audio injection API from having to manually
* unregister the policy that was used to create the track.
*/
private void unregisterAudioPolicyOnRelease(AudioPolicy audioPolicy) {
mAudioPolicy = audioPolicy;
}
/**
* 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
* {@link 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.
* @param delayInFrames number of frames to be ignored at the beginning of the stream. A value
* of 0 indicates no delay is to be applied.
* @param paddingInFrames 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.
*/
public void setOffloadDelayPadding(@IntRange(from = 0) int delayInFrames,
@IntRange(from = 0) int paddingInFrames) {
if (paddingInFrames < 0) {
throw new IllegalArgumentException("Illegal negative padding");
}
if (delayInFrames < 0) {
throw new IllegalArgumentException("Illegal negative delay");
}
if (!mOffloaded) {
throw new IllegalStateException("Illegal use of delay/padding on non-offloaded track");
}
if (mState == STATE_UNINITIALIZED) {
throw new IllegalStateException("Uninitialized track");
}
mOffloadDelayFrames = delayInFrames;
mOffloadPaddingFrames = paddingInFrames;
native_set_delay_padding(delayInFrames, paddingInFrames);
}
/**
* Return the decoder delay of an offloaded track, expressed in frames, previously set with
* {@link #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
* {@link AudioTrack.Builder#setOffloadedPlayback(boolean)}.
* @return decoder delay expressed in frames.
*/
public @IntRange(from = 0) int getOffloadDelay() {
if (!mOffloaded) {
throw new IllegalStateException("Illegal query of delay on non-offloaded track");
}
if (mState == STATE_UNINITIALIZED) {
throw new IllegalStateException("Illegal query of delay on uninitialized track");
}
return mOffloadDelayFrames;
}
/**
* Return the decoder padding of an offloaded track, expressed in frames, previously set with
* {@link #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
* {@link AudioTrack.Builder#setOffloadedPlayback(boolean)}.
* @return decoder padding expressed in frames.
*/
public @IntRange(from = 0) int getOffloadPadding() {
if (!mOffloaded) {
throw new IllegalStateException("Illegal query of padding on non-offloaded track");
}
if (mState == STATE_UNINITIALIZED) {
throw new IllegalStateException("Illegal query of padding on uninitialized track");
}
return mOffloadPaddingFrames;
}
/**
* 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 {@link #PLAYSTATE_PLAYING}
* {@see #getPlayState()}.
* Use this method in the same thread as any write() operation.
*/
public void setOffloadEndOfStream() {
if (!mOffloaded) {
throw new IllegalStateException("EOS not supported on non-offloaded track");
}
if (mState == STATE_UNINITIALIZED) {
throw new IllegalStateException("Uninitialized track");
}
if (mPlayState != PLAYSTATE_PLAYING) {
throw new IllegalStateException("EOS not supported if not playing");
}
synchronized (mStreamEventCbLock) {
if (mStreamEventCbInfoList.size() == 0) {
throw new IllegalStateException("EOS not supported without StreamEventCallback");
}
}
synchronized (mPlayStateLock) {
native_stop();
mOffloadEosPending = true;
mPlayState = PLAYSTATE_STOPPING;
}
}
/**
* Returns whether the track was built with {@link Builder#setOffloadedPlayback(boolean)} set
* to {@code true}.
* @return true if the track is using offloaded playback.
*/
public boolean isOffloadedPlayback() {
return mOffloaded;
}
/**
* 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.
* @param format a non-null {@link AudioFormat} instance describing the format of
* the audio data.
* @param attributes a non-null {@link AudioAttributes} instance.
* @return true if the given audio format can be played directly.
* @deprecated Use {@link AudioManager#getDirectPlaybackSupport(AudioFormat, AudioAttributes)}
* instead.
*/
@Deprecated
public static boolean isDirectPlaybackSupported(@NonNull AudioFormat format,
@NonNull AudioAttributes attributes) {
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat argument");
}
if (attributes == null) {
throw new IllegalArgumentException("Illegal null AudioAttributes argument");
}
return native_is_direct_output_supported(format.getEncoding(), format.getSampleRate(),
format.getChannelMask(), format.getChannelIndexMask(),
attributes.getContentType(), attributes.getUsage(), attributes.getFlags());
}
/*
* The MAX_LEVEL should be exactly representable by an IEEE 754-2008 base32 float.
* This means fractions must be divisible by a power of 2. For example,
* 10.25f is OK as 0.25 is 1/4, but 10.1f is NOT OK as 1/10 is not expressable by
* a finite binary fraction.
*
* 48.f is the nominal max for API level {@link android os.Build.VERSION_CODES#R}.
* We use this to suggest a baseline range for implementation.
*
* The API contract specification allows increasing this value in a future
* API release, but not decreasing this value.
*/
private static final float MAX_AUDIO_DESCRIPTION_MIX_LEVEL = 48.f;
private static boolean isValidAudioDescriptionMixLevel(float level) {
return !(Float.isNaN(level) || level > MAX_AUDIO_DESCRIPTION_MIX_LEVEL);
}
/**
* Sets the Audio Description mix level in dB.
*
* For AudioTracks incorporating a secondary Audio Description stream
* (where such contents may be sent through an Encapsulation Mode
* other than {@link #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.
*
* @param level a floating point value between
* {@code Float.NEGATIVE_INFINITY} to {@code +48.f},
* where {@code Float.NEGATIVE_INFINITY} means the Audio Description is not mixed
* and a level of {@code 0.f} means the Audio Description is mixed without scaling.
* @return true on success, false on failure.
*/
public boolean setAudioDescriptionMixLeveldB(
@FloatRange(to = 48.f, toInclusive = true) float level) {
if (!isValidAudioDescriptionMixLevel(level)) {
throw new IllegalArgumentException("level is out of range" + level);
}
return native_set_audio_description_mix_level_db(level) == SUCCESS;
}
/**
* Returns the Audio Description mix level in dB.
*
* If Audio Description mixing is unavailable from the hardware device,
* a value of {@code Float.NEGATIVE_INFINITY} is returned.
*
* @return the current Audio Description Mix Level in dB.
* A value of {@code 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 {@code Float.NaN}.
*/
public float getAudioDescriptionMixLeveldB() {
float[] level = { Float.NEGATIVE_INFINITY };
try {
final int status = native_get_audio_description_mix_level_db(level);
if (status != SUCCESS || Float.isNaN(level[0])) {
return Float.NEGATIVE_INFINITY;
}
} catch (Exception e) {
return Float.NEGATIVE_INFINITY;
}
return level[0];
}
private static boolean isValidDualMonoMode(@DualMonoMode int dualMonoMode) {
switch (dualMonoMode) {
case DUAL_MONO_MODE_OFF:
case DUAL_MONO_MODE_LR:
case DUAL_MONO_MODE_LL:
case DUAL_MONO_MODE_RR:
return true;
default:
return false;
}
}
/**
* 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.
*
* @param dualMonoMode one of {@link #DUAL_MONO_MODE_OFF},
* {@link #DUAL_MONO_MODE_LR},
* {@link #DUAL_MONO_MODE_LL},
* {@link #DUAL_MONO_MODE_RR}.
*
* @return true on success, false on failure if the output device
* does not support Dual Mono mode.
*/
public boolean setDualMonoMode(@DualMonoMode int dualMonoMode) {
if (!isValidDualMonoMode(dualMonoMode)) {
throw new IllegalArgumentException(
"Invalid Dual Mono mode " + dualMonoMode);
}
return native_set_dual_mono_mode(dualMonoMode) == SUCCESS;
}
/**
* Returns the Dual Mono mode presentation setting.
*
* If no Dual Mono presentation is available for the output device,
* then {@link #DUAL_MONO_MODE_OFF} is returned.
*
* @return one of {@link #DUAL_MONO_MODE_OFF},
* {@link #DUAL_MONO_MODE_LR},
* {@link #DUAL_MONO_MODE_LL},
* {@link #DUAL_MONO_MODE_RR}.
*/
public @DualMonoMode int getDualMonoMode() {
int[] dualMonoMode = { DUAL_MONO_MODE_OFF };
try {
final int status = native_get_dual_mono_mode(dualMonoMode);
if (status != SUCCESS || !isValidDualMonoMode(dualMonoMode[0])) {
return DUAL_MONO_MODE_OFF;
}
} catch (Exception e) {
return DUAL_MONO_MODE_OFF;
}
return dualMonoMode[0];
}
// mask of all the positional channels supported, however the allowed combinations
// are further restricted by the matching left/right rule and
// AudioSystem.OUT_CHANNEL_COUNT_MAX
private static final int SUPPORTED_OUT_CHANNELS =
AudioFormat.CHANNEL_OUT_FRONT_LEFT |
AudioFormat.CHANNEL_OUT_FRONT_RIGHT |
AudioFormat.CHANNEL_OUT_FRONT_CENTER |
AudioFormat.CHANNEL_OUT_LOW_FREQUENCY |
AudioFormat.CHANNEL_OUT_BACK_LEFT |
AudioFormat.CHANNEL_OUT_BACK_RIGHT |
AudioFormat.CHANNEL_OUT_FRONT_LEFT_OF_CENTER |
AudioFormat.CHANNEL_OUT_FRONT_RIGHT_OF_CENTER |
AudioFormat.CHANNEL_OUT_BACK_CENTER |
AudioFormat.CHANNEL_OUT_SIDE_LEFT |
AudioFormat.CHANNEL_OUT_SIDE_RIGHT |
AudioFormat.CHANNEL_OUT_TOP_CENTER |
AudioFormat.CHANNEL_OUT_TOP_FRONT_LEFT |
AudioFormat.CHANNEL_OUT_TOP_FRONT_CENTER |
AudioFormat.CHANNEL_OUT_TOP_FRONT_RIGHT |
AudioFormat.CHANNEL_OUT_TOP_BACK_LEFT |
AudioFormat.CHANNEL_OUT_TOP_BACK_CENTER |
AudioFormat.CHANNEL_OUT_TOP_BACK_RIGHT |
AudioFormat.CHANNEL_OUT_TOP_SIDE_LEFT |
AudioFormat.CHANNEL_OUT_TOP_SIDE_RIGHT |
AudioFormat.CHANNEL_OUT_BOTTOM_FRONT_LEFT |
AudioFormat.CHANNEL_OUT_BOTTOM_FRONT_CENTER |
AudioFormat.CHANNEL_OUT_BOTTOM_FRONT_RIGHT |
AudioFormat.CHANNEL_OUT_LOW_FREQUENCY_2 |
AudioFormat.CHANNEL_OUT_FRONT_WIDE_LEFT |
AudioFormat.CHANNEL_OUT_FRONT_WIDE_RIGHT;
// Returns a boolean whether the attributes, format, bufferSizeInBytes, mode allow
// power saving to be automatically enabled for an AudioTrack. Returns false if
// power saving is already enabled in the attributes parameter.
private static boolean shouldEnablePowerSaving(
@Nullable AudioAttributes attributes, @Nullable AudioFormat format,
int bufferSizeInBytes, int mode) {
// If no attributes, OK
// otherwise check attributes for USAGE_MEDIA and CONTENT_UNKNOWN, MUSIC, or MOVIE.
// Only consider flags that are not compatible with FLAG_DEEP_BUFFER. We include
// FLAG_DEEP_BUFFER because if set the request is explicit and
// shouldEnablePowerSaving() should return false.
final int flags = attributes.getAllFlags()
& (AudioAttributes.FLAG_DEEP_BUFFER | AudioAttributes.FLAG_LOW_LATENCY
| AudioAttributes.FLAG_HW_AV_SYNC | AudioAttributes.FLAG_BEACON);
if (attributes != null &&
(flags != 0 // cannot have any special flags
|| attributes.getUsage() != AudioAttributes.USAGE_MEDIA
|| (attributes.getContentType() != AudioAttributes.CONTENT_TYPE_UNKNOWN
&& attributes.getContentType() != AudioAttributes.CONTENT_TYPE_MUSIC
&& attributes.getContentType() != AudioAttributes.CONTENT_TYPE_MOVIE))) {
return false;
}
// Format must be fully specified and be linear pcm
if (format == null
|| format.getSampleRate() == AudioFormat.SAMPLE_RATE_UNSPECIFIED
|| !AudioFormat.isEncodingLinearPcm(format.getEncoding())
|| !AudioFormat.isValidEncoding(format.getEncoding())
|| format.getChannelCount() < 1) {
return false;
}
// Mode must be streaming
if (mode != MODE_STREAM) {
return false;
}
// A buffer size of 0 is always compatible with deep buffer (when called from the Builder)
// but for app compatibility we only use deep buffer power saving for large buffer sizes.
if (bufferSizeInBytes != 0) {
final long BUFFER_TARGET_MODE_STREAM_MS = 100;
final int MILLIS_PER_SECOND = 1000;
final long bufferTargetSize =
BUFFER_TARGET_MODE_STREAM_MS
* format.getChannelCount()
* format.getBytesPerSample(format.getEncoding())
* format.getSampleRate()
/ MILLIS_PER_SECOND;
if (bufferSizeInBytes < bufferTargetSize) {
return false;
}
}
return true;
}
// Convenience method for the constructor's parameter checks.
// This is where constructor IllegalArgumentException-s are thrown
// postconditions:
// mChannelCount is valid
// mChannelMask is valid
// mAudioFormat is valid
// mSampleRate is valid
// mDataLoadMode is valid
private void audioParamCheck(int sampleRateInHz, int channelConfig, int channelIndexMask,
int audioFormat, int mode) {
//--------------
// sample rate, note these values are subject to change
if ((sampleRateInHz < AudioFormat.SAMPLE_RATE_HZ_MIN ||
sampleRateInHz > AudioFormat.SAMPLE_RATE_HZ_MAX) &&
sampleRateInHz != AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
throw new IllegalArgumentException(sampleRateInHz
+ "Hz is not a supported sample rate.");
}
mSampleRate = sampleRateInHz;
if (audioFormat == AudioFormat.ENCODING_IEC61937
&& channelConfig != AudioFormat.CHANNEL_OUT_STEREO
&& AudioFormat.channelCountFromOutChannelMask(channelConfig) != 8) {
Log.w(TAG, "ENCODING_IEC61937 is configured with channel mask as " + channelConfig
+ ", which is not 2 or 8 channels");
}
//--------------
// channel config
mChannelConfiguration = channelConfig;
switch (channelConfig) {
case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
mChannelCount = 1;
mChannelMask = AudioFormat.CHANNEL_OUT_MONO;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
mChannelCount = 2;
mChannelMask = AudioFormat.CHANNEL_OUT_STEREO;
break;
default:
if (channelConfig == AudioFormat.CHANNEL_INVALID && channelIndexMask != 0) {
mChannelCount = 0;
break; // channel index configuration only
}
if (!isMultichannelConfigSupported(channelConfig, audioFormat)) {
throw new IllegalArgumentException(
"Unsupported channel mask configuration " + channelConfig
+ " for encoding " + audioFormat);
}
mChannelMask = channelConfig;
mChannelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig);
}
// check the channel index configuration (if present)
mChannelIndexMask = channelIndexMask;
if (mChannelIndexMask != 0) {
// As of S, we accept up to 24 channel index mask.
final int fullIndexMask = (1 << AudioSystem.FCC_24) - 1;
final int channelIndexCount = Integer.bitCount(channelIndexMask);
final boolean accepted = (channelIndexMask & ~fullIndexMask) == 0
&& (!AudioFormat.isEncodingLinearFrames(audioFormat) // compressed OK
|| channelIndexCount <= AudioSystem.OUT_CHANNEL_COUNT_MAX); // PCM
if (!accepted) {
throw new IllegalArgumentException(
"Unsupported channel index mask configuration " + channelIndexMask
+ " for encoding " + audioFormat);
}
if (mChannelCount == 0) {
mChannelCount = channelIndexCount;
} else if (mChannelCount != channelIndexCount) {
throw new IllegalArgumentException("Channel count must match");
}
}
//--------------
// audio format
if (audioFormat == AudioFormat.ENCODING_DEFAULT) {
audioFormat = AudioFormat.ENCODING_PCM_16BIT;
}
if (!AudioFormat.isPublicEncoding(audioFormat)) {
throw new IllegalArgumentException("Unsupported audio encoding.");
}
mAudioFormat = audioFormat;
//--------------
// audio load mode
if (((mode != MODE_STREAM) && (mode != MODE_STATIC)) ||
((mode != MODE_STREAM) && !AudioFormat.isEncodingLinearPcm(mAudioFormat))) {
throw new IllegalArgumentException("Invalid mode.");
}
mDataLoadMode = mode;
}
// General pair map
private static final Map The word "volume" in the API name is historical; this is actually a linear gain.
* @return the minimum value, which is the constant 0.0.
*/
static public float getMinVolume() {
return GAIN_MIN;
}
/**
* 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 the maximum value, which is greater than or equal to 1.0.
*/
static public float getMaxVolume() {
return GAIN_MAX;
}
/**
* 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 {@link #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 {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED},
* then the initial sink sample rate is a route-dependent default value based on the source [sic].
*/
public int getSampleRate() {
return mSampleRate;
}
/**
* Returns the current playback sample rate rate in Hz.
*/
public int getPlaybackRate() {
return native_get_playback_rate();
}
/**
* Returns the current playback parameters.
* See {@link #setPlaybackParams(PlaybackParams)} to set playback parameters
* @return current {@link PlaybackParams}.
* @throws IllegalStateException if track is not initialized.
*/
public @NonNull PlaybackParams getPlaybackParams() {
return native_get_playback_params();
}
/**
* Returns the {@link AudioAttributes} used in configuration.
* If a {@code streamType} is used instead of an {@code AudioAttributes}
* to configure the AudioTrack
* (the use of {@code streamType} for configuration is deprecated),
* then the {@code AudioAttributes}
* equivalent to the {@code streamType} is returned.
* @return The {@code AudioAttributes} used to configure the AudioTrack.
* @throws IllegalStateException If the track is not initialized.
*/
public @NonNull AudioAttributes getAudioAttributes() {
if (mState == STATE_UNINITIALIZED || mConfiguredAudioAttributes == null) {
throw new IllegalStateException("track not initialized");
}
return mConfiguredAudioAttributes;
}
/**
* Returns the configured audio data encoding. See {@link AudioFormat#ENCODING_PCM_8BIT},
* {@link AudioFormat#ENCODING_PCM_16BIT}, and {@link AudioFormat#ENCODING_PCM_FLOAT}.
*/
public int getAudioFormat() {
return mAudioFormat;
}
/**
* Returns the volume stream type of this AudioTrack.
* Compare the result against {@link AudioManager#STREAM_VOICE_CALL},
* {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING},
* {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM},
* {@link AudioManager#STREAM_NOTIFICATION}, {@link AudioManager#STREAM_DTMF} or
* {@link AudioManager#STREAM_ACCESSIBILITY}.
*/
public int getStreamType() {
return mStreamType;
}
/**
* Returns the configured channel position mask.
* For example, refer to {@link AudioFormat#CHANNEL_OUT_MONO},
* {@link AudioFormat#CHANNEL_OUT_STEREO}, {@link AudioFormat#CHANNEL_OUT_5POINT1}.
* This method may return {@link AudioFormat#CHANNEL_INVALID} if
* a channel index mask was used. Consider
* {@link #getFormat()} instead, to obtain an {@link AudioFormat},
* which contains both the channel position mask and the channel index mask.
*/
public int getChannelConfiguration() {
return mChannelConfiguration;
}
/**
* Returns the configured This will be less than or equal to the result of
* {@link #getBufferCapacityInFrames()}.
* It will be equal if {@link #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 See also {@link AudioManager#getProperty(String)} for key
* {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}.
* @return current size in frames of the 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
* {@link #getBufferCapacityInFrames()}.
* It may also be adjusted slightly for internal reasons.
* If bufferSizeInFrames is less than zero then {@link #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.
*
* @param bufferSizeInFrames requested buffer size in frames
* @return the actual buffer size in frames or an error code,
* {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}
* @throws IllegalStateException if track is not initialized.
*/
public int setBufferSizeInFrames(@IntRange (from = 0) int bufferSizeInFrames) {
if (mDataLoadMode == MODE_STATIC || mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
if (bufferSizeInFrames < 0) {
return ERROR_BAD_VALUE;
}
return native_set_buffer_size_frames(bufferSizeInFrames);
}
/**
* Returns the maximum size of the If the track's creation mode is {@link #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 {@link #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 See also {@link AudioManager#getProperty(String)} for key
* {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}.
* @return maximum size in frames of the The streaming start threshold is the buffer level that the written audio
* data must reach for audio streaming to start after {@link #play()} is called.
* For compressed streams, the size of a frame is considered to be exactly one byte.
*
* @param startThresholdInFrames the desired start threshold.
* @return the actual start threshold in frames value. This is
* an integer between 1 to the buffer capacity
* (see {@link #getBufferCapacityInFrames()}),
* and might change if the output sink changes after track creation.
* @throws IllegalStateException if the track is not initialized or the
* track transfer mode is not {@link #MODE_STREAM}.
* @throws IllegalArgumentException if startThresholdInFrames is not positive.
* @see #getStartThresholdInFrames()
*/
public @IntRange(from = 1) int setStartThresholdInFrames(
@IntRange (from = 1) int startThresholdInFrames) {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("AudioTrack is not initialized");
}
if (mDataLoadMode != MODE_STREAM) {
throw new IllegalStateException("AudioTrack must be a streaming track");
}
if (startThresholdInFrames < 1) {
throw new IllegalArgumentException("startThresholdInFrames "
+ startThresholdInFrames + " must be positive");
}
return native_setStartThresholdInFrames(startThresholdInFrames);
}
/**
* Returns the streaming start threshold of the The streaming start threshold is the buffer level that the written audio
* data must reach for audio streaming to start after {@link #play()} is called.
* When an For compressed streams, the size of a frame is considered to be exactly one byte.
*
* @return the current start threshold in frames value. This is
* an integer between 1 to the buffer capacity
* (see {@link #getBufferCapacityInFrames()}),
* and might change if the output sink changes after track creation.
* @throws IllegalStateException if the track is not initialized or the
* track is not {@link #MODE_STREAM}.
* @see #setStartThresholdInFrames(int)
*/
public @IntRange (from = 1) int getStartThresholdInFrames() {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("AudioTrack is not initialized");
}
if (mDataLoadMode != MODE_STREAM) {
throw new IllegalStateException("AudioTrack must be a streaming track");
}
return native_getStartThresholdInFrames();
}
/**
* Returns the frame count of the native
* 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 {@link #setBufferSizeInFrames(int)} to dynamically increase the
* effective size of the buffer.
*/
public int getUnderrunCount() {
return native_get_underrun_count();
}
/**
* Returns the current performance mode of the {@link AudioTrack}.
*
* @return one of {@link AudioTrack#PERFORMANCE_MODE_NONE},
* {@link AudioTrack#PERFORMANCE_MODE_LOW_LATENCY},
* or {@link AudioTrack#PERFORMANCE_MODE_POWER_SAVING}.
* Use {@link AudioTrack.Builder#setPerformanceMode}
* in the {@link AudioTrack.Builder} to enable a performance mode.
* @throws IllegalStateException if track is not initialized.
*/
public @PerformanceMode int getPerformanceMode() {
final int flags = native_get_flags();
if ((flags & AUDIO_OUTPUT_FLAG_FAST) != 0) {
return PERFORMANCE_MODE_LOW_LATENCY;
} else if ((flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
return PERFORMANCE_MODE_POWER_SAVING;
} else {
return PERFORMANCE_MODE_NONE;
}
}
/**
* Returns the output sample rate in Hz for the specified stream type.
*/
static public int getNativeOutputSampleRate(int streamType) {
return native_get_output_sample_rate(streamType);
}
/**
* Returns the estimated minimum buffer size required for an AudioTrack
* object to be created in the {@link #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.
* @param sampleRateInHz the source sample rate expressed in Hz.
* {@link AudioFormat#SAMPLE_RATE_UNSPECIFIED} is not permitted.
* @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_OUT_MONO} and
* {@link AudioFormat#CHANNEL_OUT_STEREO}
* @param audioFormat the format in which the audio data is represented.
* See {@link AudioFormat#ENCODING_PCM_16BIT} and
* {@link AudioFormat#ENCODING_PCM_8BIT},
* and {@link AudioFormat#ENCODING_PCM_FLOAT}.
* @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed,
* or {@link #ERROR} if unable to query for output properties,
* or the minimum buffer size expressed in bytes.
*/
static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
int channelCount = 0;
switch(channelConfig) {
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
channelCount = 2;
break;
default:
if (!isMultichannelConfigSupported(channelConfig, audioFormat)) {
loge("getMinBufferSize(): Invalid channel configuration.");
return ERROR_BAD_VALUE;
} else {
channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig);
}
}
if (!AudioFormat.isPublicEncoding(audioFormat)) {
loge("getMinBufferSize(): Invalid audio format.");
return ERROR_BAD_VALUE;
}
// sample rate, note these values are subject to change
// Note: AudioFormat.SAMPLE_RATE_UNSPECIFIED is not allowed
if ( (sampleRateInHz < AudioFormat.SAMPLE_RATE_HZ_MIN) ||
(sampleRateInHz > AudioFormat.SAMPLE_RATE_HZ_MAX) ) {
loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate.");
return ERROR_BAD_VALUE;
}
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if (size <= 0) {
loge("getMinBufferSize(): error querying hardware");
return ERROR;
}
else {
return size;
}
}
/**
* Returns the audio session ID.
*
* @return the ID of the audio session this AudioTrack belongs to.
*/
public int getAudioSessionId() {
return mSessionId;
}
/**
* 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.
*
* @param timestamp a reference to a non-null AudioTimestamp instance allocated
* and owned by caller.
* @return 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 {@link #getPlaybackHeadPosition}.
* However, it may be useful to continue to query for
* timestamps occasionally, to recover after a route change.
*/
// Add this text when the "on new timestamp" API is added:
// Use if you need to get the most recent timestamp outside of the event callback handler.
public boolean getTimestamp(AudioTimestamp timestamp)
{
if (timestamp == null) {
throw new IllegalArgumentException();
}
// It's unfortunate, but we have to either create garbage every time or use synchronized
long[] longArray = new long[2];
int ret = native_get_timestamp(longArray);
if (ret != SUCCESS) {
return false;
}
timestamp.framePosition = longArray[0];
timestamp.nanoTime = longArray[1];
return true;
}
/**
* Poll for a timestamp on demand.
*
* Same as {@link #getTimestamp(AudioTimestamp)} but with a more useful return code.
*
* @param timestamp a reference to a non-null AudioTimestamp instance allocated
* and owned by caller.
* @return {@link #SUCCESS} if a timestamp is available
* {@link #ERROR_WOULD_BLOCK} if called in STOPPED or FLUSHED state, or if called
* immediately after start/ACTIVE, when the number of frames consumed is less than the
* overall hardware latency to physical output. In WOULD_BLOCK cases, one might poll
* again, or use {@link #getPlaybackHeadPosition}, or use 0 position and current time
* for the timestamp.
* {@link #ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and
* needs to be recreated.
* {@link #ERROR_INVALID_OPERATION} if current route does not support
* timestamps. In this case, the approximate frame position can be obtained
* using {@link #getPlaybackHeadPosition}.
*
* 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.
* @hide
*/
// Add this text when the "on new timestamp" API is added:
// Use if you need to get the most recent timestamp outside of the event callback handler.
public int getTimestampWithStatus(AudioTimestamp timestamp)
{
if (timestamp == null) {
throw new IllegalArgumentException();
}
// It's unfortunate, but we have to either create garbage every time or use synchronized
long[] longArray = new long[2];
int ret = native_get_timestamp(longArray);
timestamp.framePosition = longArray[0];
timestamp.nanoTime = longArray[1];
return ret;
}
/**
* Return Metrics data about the current AudioTrack instance.
*
* @return a {@link PersistableBundle} containing the set of attributes and values
* available for the media being handled by this instance of AudioTrack
* The attributes are descibed in {@link MetricsConstants}.
*
* Additional vendor-specific fields may also be present in
* the return value.
*/
public PersistableBundle getMetrics() {
PersistableBundle bundle = native_getMetrics();
return bundle;
}
private native PersistableBundle native_getMetrics();
//--------------------------------------------------------------------------
// Initialization / configuration
//--------------------
/**
* 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.
* @param listener
*/
public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) {
setPlaybackPositionUpdateListener(listener, null);
}
/**
* 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.
* @param listener
* @param handler the Handler that will receive the event notification messages.
*/
public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener,
Handler handler) {
if (listener != null) {
mEventHandlerDelegate = new NativePositionEventHandlerDelegate(this, listener, handler);
} else {
mEventHandlerDelegate = null;
}
}
private static float clampGainOrLevel(float gainOrLevel) {
if (Float.isNaN(gainOrLevel)) {
throw new IllegalArgumentException();
}
if (gainOrLevel < GAIN_MIN) {
gainOrLevel = GAIN_MIN;
} else if (gainOrLevel > GAIN_MAX) {
gainOrLevel = GAIN_MAX;
}
return gainOrLevel;
}
/**
* 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 {@link #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.
* @param leftGain output gain for the left channel.
* @param rightGain output gain for the right channel
* @return error code or success, see {@link #SUCCESS},
* {@link #ERROR_INVALID_OPERATION}
* @deprecated Applications should use {@link #setVolume} instead, as it
* more gracefully scales down to mono, and up to multi-channel content beyond stereo.
*/
@Deprecated
public int setStereoVolume(float leftGain, float rightGain) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
baseSetVolume(leftGain, rightGain);
return SUCCESS;
}
@Override
void playerSetVolume(boolean muting, float leftVolume, float rightVolume) {
leftVolume = clampGainOrLevel(muting ? 0.0f : leftVolume);
rightVolume = clampGainOrLevel(muting ? 0.0f : rightVolume);
native_setVolume(leftVolume, rightVolume);
}
/**
* 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 {@link #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 {@link #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.
* @param gain output gain for all channels.
* @return error code or success, see {@link #SUCCESS},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setVolume(float gain) {
return setStereoVolume(gain, gain);
}
@Override
/* package */ int playerApplyVolumeShaper(
@NonNull VolumeShaper.Configuration configuration,
@NonNull VolumeShaper.Operation operation) {
return native_applyVolumeShaper(configuration, operation);
}
@Override
/* package */ @Nullable VolumeShaper.State playerGetVolumeShaperState(int id) {
return native_getVolumeShaperState(id);
}
@Override
public @NonNull VolumeShaper createVolumeShaper(
@NonNull VolumeShaper.Configuration configuration) {
return new VolumeShaper(configuration, this);
}
/**
* 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
* {@link #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
* {@link #getNativeOutputSampleRate(int)}.
* Use {@link #setPlaybackParams(PlaybackParams)} for speed control.
* This method may also be used to repurpose an existing
* If track's creation mode is {@link #MODE_STATIC}, you must have called one of
* the write methods ({@link #write(byte[], int, int)}, {@link #write(byte[], int, int, int)},
* {@link #write(short[], int, int)}, {@link #write(short[], int, int, int)},
* {@link #write(float[], int, int, int)}, or {@link #write(ByteBuffer, int, int)}) prior to
* play().
*
* If the mode is {@link #MODE_STREAM}, you can optionally prime the data path prior to
* calling play(), by writing up to
* As of {@link android.os.Build.VERSION_CODES#S} the minimum level to start playing
* can be obtained using {@link #getStartThresholdInFrames()} and set with
* {@link #setStartThresholdInFrames(int)}.
*
* @throws IllegalStateException if the track isn't properly initialized
*/
public void play()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("play() called on uninitialized AudioTrack.");
}
//FIXME use lambda to pass startImpl to superclass
final int delay = getStartDelayMs();
if (delay == 0) {
startImpl();
} else {
new Thread() {
public void run() {
try {
Thread.sleep(delay);
} catch (InterruptedException e) {
e.printStackTrace();
}
baseSetStartDelayMs(0);
try {
startImpl();
} catch (IllegalStateException e) {
// fail silently for a state exception when it is happening after
// a delayed start, as the player state could have changed between the
// call to start() and the execution of startImpl()
}
}
}.start();
}
}
private void startImpl() {
synchronized (mRoutingChangeListeners) {
if (!mEnableSelfRoutingMonitor) {
mEnableSelfRoutingMonitor = testEnableNativeRoutingCallbacksLocked();
}
}
synchronized(mPlayStateLock) {
baseStart(0); // unknown device at this point
native_start();
// FIXME see b/179218630
//baseStart(native_getRoutedDeviceId());
if (mPlayState == PLAYSTATE_PAUSED_STOPPING) {
mPlayState = PLAYSTATE_STOPPING;
} else {
mPlayState = PLAYSTATE_PLAYING;
mOffloadEosPending = false;
}
}
}
/**
* Stops playing the audio data.
* When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing
* after the last buffer that was written has been played. For an immediate stop, use
* {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played
* back yet.
* @throws IllegalStateException
*/
public void stop()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("stop() called on uninitialized AudioTrack.");
}
// stop playing
synchronized(mPlayStateLock) {
native_stop();
baseStop();
if (mOffloaded && mPlayState != PLAYSTATE_PAUSED_STOPPING) {
mPlayState = PLAYSTATE_STOPPING;
} else {
mPlayState = PLAYSTATE_STOPPED;
mOffloadEosPending = false;
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
mPlayStateLock.notify();
}
}
tryToDisableNativeRoutingCallback();
}
/**
* Pauses the playback of the audio data. Data that has not been played
* back will not be discarded. Subsequent calls to {@link #play} will play
* this data back. See {@link #flush()} to discard this data.
*
* @throws IllegalStateException
*/
public void pause()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("pause() called on uninitialized AudioTrack.");
}
// pause playback
synchronized(mPlayStateLock) {
native_pause();
basePause();
if (mPlayState == PLAYSTATE_STOPPING) {
mPlayState = PLAYSTATE_PAUSED_STOPPING;
} else {
mPlayState = PLAYSTATE_PAUSED;
}
}
}
//---------------------------------------------------------
// Audio data supply
//--------------------
/**
* 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 {@link #MODE_STREAM}.
*
* 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.
*
* @param audioData the array that holds the data to play.
* @param offsetInBytes 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.
* @param sizeInBytes 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 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.
*
* In streaming mode, the blocking behavior depends on the write mode. If the write mode is
* {@link #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
* {@link #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.
*
* @param audioData the array that holds the data to play.
* @param offsetInBytes 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.
* @param sizeInBytes 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.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no
* effect in static mode.
*
* 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.
*
* @param audioData the array that holds the data to play.
* @param offsetInShorts 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.
* @param sizeInShorts 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 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.
*
* In streaming mode, the blocking behavior depends on the write mode. If the write mode is
* {@link #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
* {@link #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.
*
* @param audioData the array that holds the data to write.
* @param offsetInShorts 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.
* @param sizeInShorts 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.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no
* effect in static mode.
*
* In streaming mode, the blocking behavior depends on the write mode. If the write mode is
* {@link #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
* {@link #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.
*
* @param audioData 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.
* @param offsetInFloats 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.
* @param sizeInFloats 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.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no
* effect in static mode.
*
* In streaming mode, the blocking behavior depends on the write mode. If the write mode is
* {@link #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
* {@link #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.
*
* @param audioData the buffer that holds the data to write, starting at the position reported
* by
* As of {@link android.os.Build.VERSION_CODES#M}, also resets the value returned by
* {@link #getPlaybackHeadPosition()} to zero.
* For earlier API levels, the reset behavior is unspecified.
*
* Use {@link #setPlaybackHeadPosition(int)} with a zero position
* if the reset of After creating an auxiliary effect (e.g.
* {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with
* {@link 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.
*
* @param effectId system wide unique id of the effect to attach
* @return error code or success, see {@link #SUCCESS},
* {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE}
*/
public int attachAuxEffect(int effectId) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
return native_attachAuxEffect(effectId);
}
/**
* Sets the send level of the audio track to the attached auxiliary effect
* {@link #attachAuxEffect(int)}. Effect levels
* are clamped to the closed interval [0.0, max] where
* max is the value of {@link #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)
*
* @param level linear send level
* @return error code or success, see {@link #SUCCESS},
* {@link #ERROR_INVALID_OPERATION}, {@link #ERROR}
*/
public int setAuxEffectSendLevel(@FloatRange(from = 0.0) float level) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
return baseSetAuxEffectSendLevel(level);
}
@Override
int playerSetAuxEffectSendLevel(boolean muting, float level) {
level = clampGainOrLevel(muting ? 0.0f : level);
int err = native_setAuxEffectSendLevel(level);
return err == 0 ? SUCCESS : ERROR;
}
//--------------------------------------------------------------------------
// Explicit Routing
//--------------------
private AudioDeviceInfo mPreferredDevice = null;
/**
* Specifies an audio device (via an {@link AudioDeviceInfo} object) to route
* the output from this AudioTrack.
* @param deviceInfo The {@link AudioDeviceInfo} specifying the audio sink.
* If deviceInfo is null, default routing is restored.
* @return true if succesful, false if the specified {@link AudioDeviceInfo} is non-null and
* does not correspond to a valid audio output device.
*/
@Override
public boolean setPreferredDevice(AudioDeviceInfo deviceInfo) {
// Do some validation....
if (deviceInfo != null && !deviceInfo.isSink()) {
return false;
}
int preferredDeviceId = deviceInfo != null ? deviceInfo.getId() : 0;
boolean status = native_setOutputDevice(preferredDeviceId);
if (status == true) {
synchronized (this) {
mPreferredDevice = deviceInfo;
}
}
return status;
}
/**
* Returns the selected output specified by {@link #setPreferredDevice}. Note that this
* is not guaranteed to correspond to the actual device being used for playback.
*/
@Override
public AudioDeviceInfo getPreferredDevice() {
synchronized (this) {
return mPreferredDevice;
}
}
/**
* Returns an {@link AudioDeviceInfo} identifying the current routing of this AudioTrack.
* Note: The query is only valid if the AudioTrack is currently playing. If it is not,
* AudioTrack to satisfy the application's
* latency requirements.
* If bufferSizeInBytes is less than the
* minimum buffer size for the output sink, it is increased to the minimum
* buffer size.
* The method {@link #getBufferSizeInFrames()} returns the
* actual size in frames of the buffer created, which
* determines the minimum frequency to write
* to the streaming AudioTrack to avoid underrun.
* See {@link #getMinBufferSize(int, int, int)} to determine the estimated minimum buffer size
* for an AudioTrack instance in streaming mode.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}.
* @param sessionId ID of audio session the AudioTrack must be attached to, or
* {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction
* time. See also {@link AudioManager#generateAudioSessionId()} to obtain a session ID before
* construction.
* @throws IllegalArgumentException
*/
public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
int mode, int sessionId)
throws IllegalArgumentException {
this(null /* context */, attributes, format, bufferSizeInBytes, mode, sessionId,
false /*offload*/, ENCAPSULATION_MODE_NONE, null /* tunerConfiguration */);
}
private AudioTrack(@Nullable Context context, AudioAttributes attributes, AudioFormat format,
int bufferSizeInBytes, int mode, int sessionId, boolean offload, int encapsulationMode,
@Nullable TunerConfiguration tunerConfiguration)
throws IllegalArgumentException {
super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
// mState already == STATE_UNINITIALIZED
mConfiguredAudioAttributes = attributes; // object copy not needed, immutable.
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat");
}
// Check if we should enable deep buffer mode
if (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {
mAttributes = new AudioAttributes.Builder(mAttributes)
.replaceFlags((mAttributes.getAllFlags()
| AudioAttributes.FLAG_DEEP_BUFFER)
& ~AudioAttributes.FLAG_LOW_LATENCY)
.build();
}
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
int rate = format.getSampleRate();
if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
rate = 0;
}
int channelIndexMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
channelIndexMask = format.getChannelIndexMask();
}
int channelMask = 0;
if ((format.getPropertySetMask()
& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
channelMask = format.getChannelMask();
} else if (channelIndexMask == 0) { // if no masks at all, use stereo
channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT
| AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
}
int encoding = AudioFormat.ENCODING_DEFAULT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
encoding = format.getEncoding();
}
audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);
mOffloaded = offload;
mStreamType = AudioSystem.STREAM_DEFAULT;
audioBuffSizeCheck(bufferSizeInBytes);
mInitializationLooper = looper;
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
int[] sampleRate = new int[] {mSampleRate};
int[] session = new int[1];
session[0] = resolvePlaybackSessionId(context, sessionId);
AttributionSource attributionSource = context == null
? AttributionSource.myAttributionSource() : context.getAttributionSource();
// native initialization
try (ScopedParcelState attributionSourceState = attributionSource.asScopedParcelState()) {
int initResult = native_setup(new WeakReferenceAudioTrack instance. By setting audio
* attributes and audio format parameters, you indicate which of those vary from the default
* behavior on the device.
* Builder is used to specify all {@link AudioFormat}
* parameters, to be used by a new AudioTrack instance:
*
*
* AudioTrack player = new AudioTrack.Builder()
* .setAudioAttributes(new AudioAttributes.Builder()
* .setUsage(AudioAttributes.USAGE_ALARM)
* .setContentType(AudioAttributes.CONTENT_TYPE_MUSIC)
* .build())
* .setAudioFormat(new AudioFormat.Builder()
* .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
* .setSampleRate(44100)
* .setChannelMask(AudioFormat.CHANNEL_OUT_STEREO)
* .build())
* .setBufferSizeInBytes(minBuffSize)
* .build();
*
*
If the audio format is not specified or is incomplete, its channel configuration will be
* {@link AudioFormat#CHANNEL_OUT_STEREO} and the encoding will be
* {@link AudioFormat#ENCODING_PCM_16BIT}.
* The sample rate will depend on the device actually selected for playback and can be queried
* with {@link #getSampleRate()} method.
*
If the buffer size is not specified with {@link #setBufferSizeInBytes(int)},
* and the mode is {@link AudioTrack#MODE_STREAM}, the minimum buffer size is used.
*
If the transfer mode is not specified with {@link #setTransferMode(int)},
* MODE_STREAM will be used.
*
If the session ID is not specified with {@link #setSessionId(int)}, a new one will
* be generated.
*
Offload is false by default.
*/
public static class Builder {
private Context mContext;
private AudioAttributes mAttributes;
private AudioFormat mFormat;
private int mBufferSizeInBytes;
private int mEncapsulationMode = ENCAPSULATION_MODE_NONE;
private int mSessionId = AUDIO_SESSION_ID_GENERATE;
private int mMode = MODE_STREAM;
private int mPerformanceMode = PERFORMANCE_MODE_NONE;
private boolean mOffload = false;
private TunerConfiguration mTunerConfiguration;
private int mCallRedirectionMode = AudioManager.CALL_REDIRECT_NONE;
/**
* Constructs a new Builder with the default values as described above.
*/
public Builder() {
}
/**
* Sets the context the track belongs to. This context will be used to pull information,
* such as {@link android.content.AttributionSource} and device specific audio session ids,
* which will be associated with the {@link AudioTrack}. However, the context itself will
* not be retained by the {@link AudioTrack}.
* @param context a non-null {@link Context} instance
* @return the same Builder instance.
*/
public @NonNull Builder setContext(@NonNull Context context) {
mContext = Objects.requireNonNull(context);
return this;
}
/**
* Sets the {@link AudioAttributes}.
* @param attributes a non-null {@link AudioAttributes} instance that describes the audio
* data to be played.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public @NonNull Builder setAudioAttributes(@NonNull AudioAttributes attributes)
throws IllegalArgumentException {
if (attributes == null) {
throw new IllegalArgumentException("Illegal null AudioAttributes argument");
}
// keep reference, we only copy the data when building
mAttributes = attributes;
return this;
}
/**
* Sets the format of the audio data to be played by the {@link AudioTrack}.
* See {@link AudioFormat.Builder} for configuring the audio format parameters such
* as encoding, channel mask and sample rate.
* @param format a non-null {@link AudioFormat} instance.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public @NonNull Builder setAudioFormat(@NonNull AudioFormat format)
throws IllegalArgumentException {
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat argument");
}
// keep reference, we only copy the data when building
mFormat = format;
return this;
}
/**
* Sets the total size (in bytes) of the buffer where audio data is read from for playback.
* If using the {@link AudioTrack} in streaming mode
* (see {@link AudioTrack#MODE_STREAM}, you can write data into this buffer in smaller
* chunks than this size. See {@link #getMinBufferSize(int, int, int)} to determine
* the estimated minimum buffer size for the creation of an AudioTrack instance
* in streaming mode.
*
If using the AudioTrack in static mode (see
* {@link AudioTrack#MODE_STATIC}), this is the maximum size of the sound that will be
* played by this instance.
* @param bufferSizeInBytes
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public @NonNull Builder setBufferSizeInBytes(@IntRange(from = 0) int bufferSizeInBytes)
throws IllegalArgumentException {
if (bufferSizeInBytes <= 0) {
throw new IllegalArgumentException("Invalid buffer size " + bufferSizeInBytes);
}
mBufferSizeInBytes = bufferSizeInBytes;
return this;
}
/**
* Sets the encapsulation mode.
*
* Encapsulation mode allows metadata to be sent together with
* the audio data payload in a {@code ByteBuffer}.
* This requires a compatible hardware audio codec.
*
* @param encapsulationMode one of {@link AudioTrack#ENCAPSULATION_MODE_NONE},
* or {@link AudioTrack#ENCAPSULATION_MODE_ELEMENTARY_STREAM}.
* @return the same Builder instance.
*/
// Note: with the correct permission {@code AudioTrack#ENCAPSULATION_MODE_HANDLE}
// may be used as well.
public @NonNull Builder setEncapsulationMode(@EncapsulationMode int encapsulationMode) {
switch (encapsulationMode) {
case ENCAPSULATION_MODE_NONE:
case ENCAPSULATION_MODE_ELEMENTARY_STREAM:
case ENCAPSULATION_MODE_HANDLE:
mEncapsulationMode = encapsulationMode;
break;
default:
throw new IllegalArgumentException(
"Invalid encapsulation mode " + encapsulationMode);
}
return this;
}
/**
* Sets the mode under which buffers of audio data are transferred from the
* {@link AudioTrack} to the framework.
* @param mode one of {@link AudioTrack#MODE_STREAM}, {@link AudioTrack#MODE_STATIC}.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public @NonNull Builder setTransferMode(@TransferMode int mode)
throws IllegalArgumentException {
switch(mode) {
case MODE_STREAM:
case MODE_STATIC:
mMode = mode;
break;
default:
throw new IllegalArgumentException("Invalid transfer mode " + mode);
}
return this;
}
/**
* Sets the session ID the {@link AudioTrack} will be attached to.
*
* Note, that if there's a device specific session id asociated with the context, explicitly
* setting a session id using this method will override it
* (see {@link Builder#setContext(Context)}).
* @param sessionId a strictly positive ID number retrieved from another
* AudioTrack via {@link AudioTrack#getAudioSessionId()} or allocated by
* {@link AudioManager} via {@link AudioManager#generateAudioSessionId()}, or
* {@link AudioManager#AUDIO_SESSION_ID_GENERATE}.
* @return the same Builder instance.
* @throws IllegalArgumentException
*/
public @NonNull Builder setSessionId(@IntRange(from = 1) int sessionId)
throws IllegalArgumentException {
if ((sessionId != AUDIO_SESSION_ID_GENERATE) && (sessionId < 1)) {
throw new IllegalArgumentException("Invalid audio session ID " + sessionId);
}
mSessionId = sessionId;
return this;
}
/**
* Sets the {@link AudioTrack} performance mode. This is an advisory request which
* may not be supported by the particular device, and the framework is free
* to ignore such request if it is incompatible with other requests or hardware.
*
* @param performanceMode one of
* {@link AudioTrack#PERFORMANCE_MODE_NONE},
* {@link AudioTrack#PERFORMANCE_MODE_LOW_LATENCY},
* or {@link AudioTrack#PERFORMANCE_MODE_POWER_SAVING}.
* @return the same Builder instance.
* @throws IllegalArgumentException if {@code performanceMode} is not valid.
*/
public @NonNull Builder setPerformanceMode(@PerformanceMode int performanceMode) {
switch (performanceMode) {
case PERFORMANCE_MODE_NONE:
case PERFORMANCE_MODE_LOW_LATENCY:
case PERFORMANCE_MODE_POWER_SAVING:
mPerformanceMode = performanceMode;
break;
default:
throw new IllegalArgumentException(
"Invalid performance mode " + performanceMode);
}
return this;
}
/**
* Sets whether this track will play through the offloaded audio path.
* When set to true, at build time, the audio format will be checked against
* {@link AudioManager#isOffloadedPlaybackSupported(AudioFormat,AudioAttributes)}
* to verify the audio format used by this track is supported on the device's offload
* path (if any).
*
Offload is only supported for media audio streams, and therefore requires that
* the usage be {@link AudioAttributes#USAGE_MEDIA}.
* @param offload true to require the offload path for playback.
* @return the same Builder instance.
*/
public @NonNull Builder setOffloadedPlayback(boolean offload) {
mOffload = offload;
return this;
}
/**
* Sets the tuner configuration for the {@code AudioTrack}.
*
* The {@link AudioTrack.TunerConfiguration} consists of parameters obtained from
* the Android TV tuner API which indicate the audio content stream id and the
* synchronization id for the {@code AudioTrack}.
*
* @param tunerConfiguration obtained by {@link AudioTrack.TunerConfiguration.Builder}.
* @return the same Builder instance.
* @hide
*/
@SystemApi
@RequiresPermission(android.Manifest.permission.MODIFY_AUDIO_ROUTING)
public @NonNull Builder setTunerConfiguration(
@NonNull TunerConfiguration tunerConfiguration) {
if (tunerConfiguration == null) {
throw new IllegalArgumentException("tunerConfiguration is null");
}
mTunerConfiguration = tunerConfiguration;
return this;
}
/**
* @hide
* Sets the {@link AudioTrack} call redirection mode.
* Used when creating an AudioTrack to inject audio to call uplink path. The mode
* indicates if the call is a PSTN call or a VoIP call in which case a dynamic audio
* policy is created to use this track as the source for all capture with voice
* communication preset.
*
* @param callRedirectionMode one of
* {@link AudioManager#CALL_REDIRECT_NONE},
* {@link AudioManager#CALL_REDIRECT_PSTN},
* or {@link AAudioManager#CALL_REDIRECT_VOIP}.
* @return the same Builder instance.
* @throws IllegalArgumentException if {@code callRedirectionMode} is not valid.
*/
public @NonNull Builder setCallRedirectionMode(
@AudioManager.CallRedirectionMode int callRedirectionMode) {
switch (callRedirectionMode) {
case AudioManager.CALL_REDIRECT_NONE:
case AudioManager.CALL_REDIRECT_PSTN:
case AudioManager.CALL_REDIRECT_VOIP:
mCallRedirectionMode = callRedirectionMode;
break;
default:
throw new IllegalArgumentException(
"Invalid call redirection mode " + callRedirectionMode);
}
return this;
}
private @NonNull AudioTrack buildCallInjectionTrack() {
AudioMixingRule audioMixingRule = new AudioMixingRule.Builder()
.addMixRule(AudioMixingRule.RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET,
new AudioAttributes.Builder()
.setCapturePreset(MediaRecorder.AudioSource.VOICE_COMMUNICATION)
.setForCallRedirection()
.build())
.setTargetMixRole(AudioMixingRule.MIX_ROLE_INJECTOR)
.build();
AudioMix audioMix = new AudioMix.Builder(audioMixingRule)
.setFormat(mFormat)
.setRouteFlags(AudioMix.ROUTE_FLAG_LOOP_BACK)
.build();
AudioPolicy audioPolicy =
new AudioPolicy.Builder(/*context=*/ mContext).addMix(audioMix).build();
if (AudioManager.registerAudioPolicyStatic(audioPolicy) != 0) {
throw new UnsupportedOperationException("Error: could not register audio policy");
}
AudioTrack track = audioPolicy.createAudioTrackSource(audioMix);
if (track == null) {
throw new UnsupportedOperationException("Cannot create injection AudioTrack");
}
track.unregisterAudioPolicyOnRelease(audioPolicy);
return track;
}
/**
* Builds an {@link AudioTrack} instance initialized with all the parameters set
* on this Builder.
* @return a new successfully initialized {@link AudioTrack} instance.
* @throws UnsupportedOperationException if the parameters set on the Builder
* were incompatible, or if they are not supported by the device,
* or if the device was not available.
*/
public @NonNull AudioTrack build() throws UnsupportedOperationException {
if (mAttributes == null) {
mAttributes = new AudioAttributes.Builder()
.setUsage(AudioAttributes.USAGE_MEDIA)
.build();
}
switch (mPerformanceMode) {
case PERFORMANCE_MODE_LOW_LATENCY:
mAttributes = new AudioAttributes.Builder(mAttributes)
.replaceFlags((mAttributes.getAllFlags()
| AudioAttributes.FLAG_LOW_LATENCY)
& ~AudioAttributes.FLAG_DEEP_BUFFER)
.build();
break;
case PERFORMANCE_MODE_NONE:
if (!shouldEnablePowerSaving(mAttributes, mFormat, mBufferSizeInBytes, mMode)) {
break; // do not enable deep buffer mode.
}
// permitted to fall through to enable deep buffer
case PERFORMANCE_MODE_POWER_SAVING:
mAttributes = new AudioAttributes.Builder(mAttributes)
.replaceFlags((mAttributes.getAllFlags()
| AudioAttributes.FLAG_DEEP_BUFFER)
& ~AudioAttributes.FLAG_LOW_LATENCY)
.build();
break;
}
if (mFormat == null) {
mFormat = new AudioFormat.Builder()
.setChannelMask(AudioFormat.CHANNEL_OUT_STEREO)
//.setSampleRate(AudioFormat.SAMPLE_RATE_UNSPECIFIED)
.setEncoding(AudioFormat.ENCODING_DEFAULT)
.build();
}
if (mCallRedirectionMode == AudioManager.CALL_REDIRECT_VOIP) {
return buildCallInjectionTrack();
} else if (mCallRedirectionMode == AudioManager.CALL_REDIRECT_PSTN) {
mAttributes = new AudioAttributes.Builder(mAttributes)
.setForCallRedirection()
.build();
}
if (mOffload) {
if (mPerformanceMode == PERFORMANCE_MODE_LOW_LATENCY) {
throw new UnsupportedOperationException(
"Offload and low latency modes are incompatible");
}
if (AudioSystem.getDirectPlaybackSupport(mFormat, mAttributes)
== AudioSystem.DIRECT_NOT_SUPPORTED) {
throw new UnsupportedOperationException(
"Cannot create AudioTrack, offload format / attributes not supported");
}
}
// TODO: Check mEncapsulationMode compatibility with MODE_STATIC, etc?
// If the buffer size is not specified in streaming mode,
// use a single frame for the buffer size and let the
// native code figure out the minimum buffer size.
if (mMode == MODE_STREAM && mBufferSizeInBytes == 0) {
int bytesPerSample = 1;
if (AudioFormat.isEncodingLinearFrames(mFormat.getEncoding())) {
try {
bytesPerSample = mFormat.getBytesPerSample(mFormat.getEncoding());
} catch (IllegalArgumentException e) {
// do nothing
}
}
mBufferSizeInBytes = mFormat.getChannelCount() * bytesPerSample;
}
try {
final AudioTrack track = new AudioTrack(
mContext, mAttributes, mFormat, mBufferSizeInBytes, mMode, mSessionId,
mOffload, mEncapsulationMode, mTunerConfiguration);
if (track.getState() == STATE_UNINITIALIZED) {
// release is not necessary
throw new UnsupportedOperationException("Cannot create AudioTrack");
}
return track;
} catch (IllegalArgumentException e) {
throw new UnsupportedOperationException(e.getMessage());
}
}
}
/**
* Sets an {@link AudioPolicy} to automatically unregister when the track is released.
*
* AudioTrack format.
* @return an {@link AudioFormat} containing the
* AudioTrack parameters at the time of configuration.
*/
public @NonNull AudioFormat getFormat() {
AudioFormat.Builder builder = new AudioFormat.Builder()
.setSampleRate(mSampleRate)
.setEncoding(mAudioFormat);
if (mChannelConfiguration != AudioFormat.CHANNEL_INVALID) {
builder.setChannelMask(mChannelConfiguration);
}
if (mChannelIndexMask != AudioFormat.CHANNEL_INVALID /* 0 */) {
builder.setChannelIndexMask(mChannelIndexMask);
}
return builder.build();
}
/**
* Returns the configured number of channels.
*/
public int getChannelCount() {
return mChannelCount;
}
/**
* 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.
* @see #STATE_UNINITIALIZED
* @see #STATE_INITIALIZED
* @see #STATE_NO_STATIC_DATA
*/
public int getState() {
return mState;
}
/**
* Returns the playback state of the AudioTrack instance.
* @see #PLAYSTATE_STOPPED
* @see #PLAYSTATE_PAUSED
* @see #PLAYSTATE_PLAYING
*/
public int getPlayState() {
synchronized (mPlayStateLock) {
switch (mPlayState) {
case PLAYSTATE_STOPPING:
return PLAYSTATE_PLAYING;
case PLAYSTATE_PAUSED_STOPPING:
return PLAYSTATE_PAUSED;
default:
return mPlayState;
}
}
}
/**
* Returns the effective size of the AudioTrack buffer
* that the application writes to.
* AudioTrack encoding indicates compressed data,
* e.g. {@link AudioFormat#ENCODING_AC3}, then the frame count returned is
* the size of the AudioTrack buffer in bytes.
* AudioTrack buffer.
* @throws IllegalStateException if track is not initialized.
*/
public @IntRange (from = 0) int getBufferSizeInFrames() {
return native_get_buffer_size_frames();
}
/**
* Limits the effective size of the AudioTrack buffer
* that the application writes to.
* AudioTrack buffer in frames.
* AudioTrack encoding indicates compressed data,
* e.g. {@link AudioFormat#ENCODING_AC3}, then the frame count returned is
* the size of the AudioTrack buffer in bytes.
* AudioTrack buffer.
* @throws IllegalStateException if track is not initialized.
*/
public @IntRange (from = 0) int getBufferCapacityInFrames() {
return native_get_buffer_capacity_frames();
}
/**
* Sets the streaming start threshold for an AudioTrack.
* AudioTrack.
* AudioTrack is created, the streaming start threshold
* is the buffer capacity in frames. If the buffer size in frames is reduced
* by {@link #setBufferSizeInFrames(int)} to a value smaller than the start threshold
* then that value will be used instead for the streaming start threshold.
* AudioTrack buffer.
* @return current size in frames of the AudioTrack buffer.
* @throws IllegalStateException
* @deprecated Use the identical public method {@link #getBufferSizeInFrames()} instead.
*/
@Deprecated
protected int getNativeFrameCount() {
return native_get_buffer_capacity_frames();
}
/**
* Returns marker position expressed in frames.
* @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition},
* or zero if marker is disabled.
*/
public int getNotificationMarkerPosition() {
return native_get_marker_pos();
}
/**
* Returns the notification update period expressed in frames.
* Zero means that no position update notifications are being delivered.
*/
public int getPositionNotificationPeriod() {
return native_get_pos_update_period();
}
/**
* 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 {@link #flush()}, {@link #reloadStaticData()}, and {@link #stop()}.
* If the track's creation mode is {@link #MODE_STATIC}, the return value indicates
* the total number of frames played since reset,
* not the current offset within the buffer.
*/
public int getPlaybackHeadPosition() {
return native_get_position();
}
/**
* Returns this track's estimated latency in milliseconds. This includes the latency due
* to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver.
*
* DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need
* a better solution.
* @hide
*/
@UnsupportedAppUsage(trackingBug = 130237544)
public int getLatency() {
return native_get_latency();
}
/**
* 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.
* AudioTrack
* for playback of content of differing sample rate,
* but with identical encoding and channel mask.
* @param sampleRateInHz the sample rate expressed in Hz
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setPlaybackRate(int sampleRateInHz) {
if (mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
}
if (sampleRateInHz <= 0) {
return ERROR_BAD_VALUE;
}
return native_set_playback_rate(sampleRateInHz);
}
/**
* 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 {@link #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
* {@link #getMinBufferSize(int, int, int)}) to allow proper playback.
* @param params see {@link PlaybackParams}. In particular,
* speed, pitch, and audio mode should be set.
* @throws IllegalArgumentException if the parameters are invalid or not accepted.
* @throws IllegalStateException if track is not initialized.
*/
public void setPlaybackParams(@NonNull PlaybackParams params) {
if (params == null) {
throw new IllegalArgumentException("params is null");
}
native_set_playback_params(params);
}
/**
* Sets the position of the notification marker. At most one marker can be active.
* @param markerInFrames marker position in wrapping frame units similar to
* {@link #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 error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setNotificationMarkerPosition(int markerInFrames) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
return native_set_marker_pos(markerInFrames);
}
/**
* Sets the period for the periodic notification event.
* @param periodInFrames update period expressed in frames.
* Zero period means no position updates. A negative period is not allowed.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION}
*/
public int setPositionNotificationPeriod(int periodInFrames) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
return native_set_pos_update_period(periodInFrames);
}
/**
* 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 {@link #MODE_STATIC} mode.
* @param positionInFrames 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 {@link #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 {@link 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.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setPlaybackHeadPosition(@IntRange (from = 0) int positionInFrames) {
if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED ||
getPlayState() == PLAYSTATE_PLAYING) {
return ERROR_INVALID_OPERATION;
}
if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) {
return ERROR_BAD_VALUE;
}
return native_set_position(positionInFrames);
}
/**
* 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 {@link #MODE_STATIC} mode.
* @param startInFrames loop start marker expressed in frames.
* Zero corresponds to start of buffer.
* The start marker must not be greater than or equal to the buffer size in frames, or negative.
* @param endInFrames loop end marker expressed in frames.
* The total buffer size in frames corresponds to end of buffer.
* The end marker must not be greater than the buffer size in frames.
* For looping, the end marker must not be less than or equal to the start marker,
* but to disable looping
* it is permitted for start marker, end marker, and loop count to all be 0.
* If any input parameters are out of range, this method returns {@link #ERROR_BAD_VALUE}.
* If the loop period (endInFrames - startInFrames) is too small for the implementation to
* support,
* {@link #ERROR_BAD_VALUE} is returned.
* The loop range is the interval [startInFrames, endInFrames).
*
* As of {@link 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.
* @param loopCount 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.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setLoopPoints(@IntRange (from = 0) int startInFrames,
@IntRange (from = 0) int endInFrames, @IntRange (from = -1) int loopCount) {
if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED ||
getPlayState() == PLAYSTATE_PLAYING) {
return ERROR_INVALID_OPERATION;
}
if (loopCount == 0) {
; // explicitly allowed as an exception to the loop region range check
} else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames &&
startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) {
return ERROR_BAD_VALUE;
}
return native_set_loop(startInFrames, endInFrames, loopCount);
}
/**
* Sets the audio presentation.
* If the audio presentation is invalid then {@link #ERROR_BAD_VALUE} will be returned.
* If a multi-stream decoder (MSD) is not present, or the format does not support
* multiple presentations, then {@link #ERROR_INVALID_OPERATION} will be returned.
* {@link #ERROR} is returned in case of any other error.
* @param presentation see {@link AudioPresentation}. In particular, id should be set.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR},
* {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}
* @throws IllegalArgumentException if the audio presentation is null.
* @throws IllegalStateException if track is not initialized.
*/
public int setPresentation(@NonNull AudioPresentation presentation) {
if (presentation == null) {
throw new IllegalArgumentException("audio presentation is null");
}
return native_setPresentation(presentation.getPresentationId(),
presentation.getProgramId());
}
/**
* 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.
* @param state the state of the AudioTrack instance
* @deprecated Only accessible by subclasses, which are not recommended for AudioTrack.
*/
@Deprecated
protected void setState(int state) {
mState = state;
}
//---------------------------------------------------------
// Transport control methods
//--------------------
/**
* Starts playing an AudioTrack.
* 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.
*
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 {@link #write(byte[], int, int)} with sizeInBytes
* less than or equal to the total buffer size
* may return a short actual transfer count.
*/
public void flush() {
if (mState == STATE_INITIALIZED) {
// flush the data in native layer
native_flush();
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
}
}
/**
* 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
* {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array.
* The format can be {@link AudioFormat#ENCODING_PCM_16BIT}, but this is deprecated.
*
*
* This is equivalent to {@link #write(byte[], int, int, int)} with writeMode
* set to {@link #WRITE_BLOCKING}.
*/
public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes) {
return write(audioData, offsetInBytes, sizeInBytes, WRITE_BLOCKING);
}
/**
* 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
* {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array.
* The format can be {@link AudioFormat#ENCODING_PCM_16BIT}, but this is deprecated.
*
With {@link #WRITE_BLOCKING}, the write will block until all data has been written
* to the audio sink.
*
With {@link #WRITE_NON_BLOCKING}, the write will return immediately after
* queuing as much audio data for playback as possible without blocking.
* @return 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.
*
*
*/
public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes,
@WriteMode int writeMode) {
// Note: we allow writes of extended integers and compressed formats from a byte array.
if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes < 0) // detect integer overflow
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
/**
* 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
* {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array.
*
*
* This is equivalent to {@link #write(short[], int, int, int)} with writeMode
* set to {@link #WRITE_BLOCKING}.
*/
public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts) {
return write(audioData, offsetInShorts, sizeInShorts, WRITE_BLOCKING);
}
/**
* 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
* {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array.
*
With {@link #WRITE_BLOCKING}, the write will block until all data has been written
* to the audio sink.
*
With {@link #WRITE_NON_BLOCKING}, the write will return immediately after
* queuing as much audio data for playback as possible without blocking.
* @return 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.
*
*
*/
public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED
|| mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT
// use ByteBuffer or byte[] instead for later encodings
|| mAudioFormat > AudioFormat.ENCODING_LEGACY_SHORT_ARRAY_THRESHOLD) {
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
|| (offsetInShorts + sizeInShorts < 0) // detect integer overflow
|| (offsetInShorts + sizeInShorts > audioData.length)) {
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
/**
* 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
* {@link AudioFormat#ENCODING_PCM_FLOAT} to correspond to the data in the array.
*
With {@link #WRITE_BLOCKING}, the write will block until all data has been written
* to the audio sink.
*
With {@link #WRITE_NON_BLOCKING}, the write will return immediately after
* queuing as much audio data for playback as possible without blocking.
* @return 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.
*
*
*/
public int write(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) {
Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0)
|| (offsetInFloats + sizeInFloats < 0) // detect integer overflow
|| (offsetInFloats + sizeInFloats > audioData.length)) {
Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
final int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat,
writeMode == WRITE_BLOCKING);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
/**
* 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.
* 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.
* @param sizeInBytes 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.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no
* effect in static mode.
*
With {@link #WRITE_BLOCKING}, the write will block until all data has been written
* to the audio sink.
*
With {@link #WRITE_NON_BLOCKING}, the write will return immediately after
* queuing as much audio data for playback as possible without blocking.
* @return zero or the positive number of bytes that were written, or one of the following
* error codes.
*
*
*/
public int write(@NonNull ByteBuffer audioData, int sizeInBytes,
@WriteMode int writeMode) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) {
Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
int ret = 0;
if (audioData.isDirect()) {
ret = native_write_native_bytes(audioData,
audioData.position(), sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
} else {
ret = native_write_byte(NioUtils.unsafeArray(audioData),
NioUtils.unsafeArrayOffset(audioData) + audioData.position(),
sizeInBytes, mAudioFormat,
writeMode == WRITE_BLOCKING);
}
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
if (ret > 0) {
audioData.position(audioData.position() + ret);
}
return ret;
}
/**
* 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.
* @param audioData 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.
* @param sizeInBytes 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.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}.
*
With {@link #WRITE_BLOCKING}, the write will block until all data has been written
* to the audio sink.
*
With {@link #WRITE_NON_BLOCKING}, the write will return immediately after
* queuing as much audio data for playback as possible without blocking.
* @param timestamp The timestamp, in nanoseconds, of the first decodable audio frame in the
* provided audioData.
* @return zero or the positive number of bytes that were written, or one of the following
* error codes.
*
*
*/
public int write(@NonNull ByteBuffer audioData, int sizeInBytes,
@WriteMode int writeMode, long timestamp) {
if (mState == STATE_UNINITIALIZED) {
Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED");
return ERROR_INVALID_OPERATION;
}
if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) {
Log.e(TAG, "AudioTrack.write() called with invalid blocking mode");
return ERROR_BAD_VALUE;
}
if (mDataLoadMode != MODE_STREAM) {
Log.e(TAG, "AudioTrack.write() with timestamp called for non-streaming mode track");
return ERROR_INVALID_OPERATION;
}
if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) == 0) {
Log.d(TAG, "AudioTrack.write() called on a regular AudioTrack. Ignoring pts...");
return write(audioData, sizeInBytes, writeMode);
}
if ((audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) {
Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value");
return ERROR_BAD_VALUE;
}
if (!blockUntilOffloadDrain(writeMode)) {
return 0;
}
// create timestamp header if none exists
if (mAvSyncHeader == null) {
mAvSyncHeader = ByteBuffer.allocate(mOffset);
mAvSyncHeader.order(ByteOrder.BIG_ENDIAN);
mAvSyncHeader.putInt(0x55550002);
}
if (mAvSyncBytesRemaining == 0) {
mAvSyncHeader.putInt(4, sizeInBytes);
mAvSyncHeader.putLong(8, timestamp);
mAvSyncHeader.putInt(16, mOffset);
mAvSyncHeader.position(0);
mAvSyncBytesRemaining = sizeInBytes;
}
// write timestamp header if not completely written already
int ret = 0;
if (mAvSyncHeader.remaining() != 0) {
ret = write(mAvSyncHeader, mAvSyncHeader.remaining(), writeMode);
if (ret < 0) {
Log.e(TAG, "AudioTrack.write() could not write timestamp header!");
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
return ret;
}
if (mAvSyncHeader.remaining() > 0) {
Log.v(TAG, "AudioTrack.write() partial timestamp header written.");
return 0;
}
}
// write audio data
int sizeToWrite = Math.min(mAvSyncBytesRemaining, sizeInBytes);
ret = write(audioData, sizeToWrite, writeMode);
if (ret < 0) {
Log.e(TAG, "AudioTrack.write() could not write audio data!");
mAvSyncHeader = null;
mAvSyncBytesRemaining = 0;
return ret;
}
mAvSyncBytesRemaining -= ret;
return ret;
}
/**
* 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 {@link #MODE_STATIC}.
* getPlaybackHeadPosition() is not needed.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int reloadStaticData() {
if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
}
return native_reload_static();
}
/**
* When an AudioTrack in offload mode is in STOPPING play state, wait until event STREAM_END is
* received if blocking write or return with 0 frames written if non blocking mode.
*/
private boolean blockUntilOffloadDrain(int writeMode) {
synchronized (mPlayStateLock) {
while (mPlayState == PLAYSTATE_STOPPING || mPlayState == PLAYSTATE_PAUSED_STOPPING) {
if (writeMode == WRITE_NON_BLOCKING) {
return false;
}
try {
mPlayStateLock.wait();
} catch (InterruptedException e) {
}
}
return true;
}
}
//--------------------------------------------------------------------------
// Audio effects management
//--------------------
/**
* 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)}.
* getRoutedDevice() will return null.
*/
@Override
public AudioDeviceInfo getRoutedDevice() {
int deviceId = native_getRoutedDeviceId();
if (deviceId == 0) {
return null;
}
return AudioManager.getDeviceForPortId(deviceId, AudioManager.GET_DEVICES_OUTPUTS);
}
private void tryToDisableNativeRoutingCallback() {
synchronized (mRoutingChangeListeners) {
if (mEnableSelfRoutingMonitor) {
mEnableSelfRoutingMonitor = false;
testDisableNativeRoutingCallbacksLocked();
}
}
}
/**
* Call BEFORE adding a routing callback handler and when enabling self routing listener
* @return returns true for success, false otherwise.
*/
@GuardedBy("mRoutingChangeListeners")
private boolean testEnableNativeRoutingCallbacksLocked() {
if (mRoutingChangeListeners.size() == 0 && !mEnableSelfRoutingMonitor) {
try {
native_enableDeviceCallback();
return true;
} catch (IllegalStateException e) {
if (Log.isLoggable(TAG, Log.DEBUG)) {
Log.d(TAG, "testEnableNativeRoutingCallbacks failed", e);
}
}
}
return false;
}
/*
* Call AFTER removing a routing callback handler and when disabling self routing listener.
*/
@GuardedBy("mRoutingChangeListeners")
private void testDisableNativeRoutingCallbacksLocked() {
if (mRoutingChangeListeners.size() == 0 && !mEnableSelfRoutingMonitor) {
try {
native_disableDeviceCallback();
} catch (IllegalStateException e) {
// Fail silently as track state could have changed in between stop
// and disabling routing callback
}
}
}
//--------------------------------------------------------------------------
// (Re)Routing Info
//--------------------
/**
* The list of AudioRouting.OnRoutingChangedListener interfaces added (with
* {@link #addOnRoutingChangedListener(android.media.AudioRouting.OnRoutingChangedListener, Handler)}
* by an app to receive (re)routing notifications.
*/
@GuardedBy("mRoutingChangeListeners")
private ArrayMapnull, the {@link Handler} associated with the main
* {@link Looper} will be used.
*/
@Override
public void addOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener,
Handler handler) {
synchronized (mRoutingChangeListeners) {
if (listener != null && !mRoutingChangeListeners.containsKey(listener)) {
mEnableSelfRoutingMonitor = testEnableNativeRoutingCallbacksLocked();
mRoutingChangeListeners.put(
listener, new NativeRoutingEventHandlerDelegate(this, listener,
handler != null ? handler : new Handler(mInitializationLooper)));
}
}
}
/**
* Removes an {@link AudioRouting.OnRoutingChangedListener} which has been previously added
* to receive rerouting notifications.
* @param listener The previously added {@link AudioRouting.OnRoutingChangedListener} interface
* to remove.
*/
@Override
public void removeOnRoutingChangedListener(AudioRouting.OnRoutingChangedListener listener) {
synchronized (mRoutingChangeListeners) {
if (mRoutingChangeListeners.containsKey(listener)) {
mRoutingChangeListeners.remove(listener);
}
testDisableNativeRoutingCallbacksLocked();
}
}
//--------------------------------------------------------------------------
// (Re)Routing Info
//--------------------
/**
* Defines the interface by which applications can receive notifications of
* routing changes for the associated {@link AudioTrack}.
*
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public interface OnRoutingChangedListener extends AudioRouting.OnRoutingChangedListener {
/**
* Called when the routing of an AudioTrack changes from either and
* explicit or policy rerouting. Use {@link #getRoutedDevice()} to
* retrieve the newly routed-to device.
*/
public void onRoutingChanged(AudioTrack audioTrack);
@Override
default public void onRoutingChanged(AudioRouting router) {
if (router instanceof AudioTrack) {
onRoutingChanged((AudioTrack) router);
}
}
}
/**
* Adds an {@link OnRoutingChangedListener} to receive notifications of routing changes
* on this AudioTrack.
* @param listener The {@link OnRoutingChangedListener} interface to receive notifications
* of rerouting events.
* @param handler Specifies the {@link Handler} object for the thread on which to execute
* the callback. If null, the {@link Handler} associated with the main
* {@link Looper} will be used.
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public void addOnRoutingChangedListener(OnRoutingChangedListener listener,
android.os.Handler handler) {
addOnRoutingChangedListener((AudioRouting.OnRoutingChangedListener) listener, handler);
}
/**
* Removes an {@link OnRoutingChangedListener} which has been previously added
* to receive rerouting notifications.
* @param listener The previously added {@link OnRoutingChangedListener} interface to remove.
* @deprecated users should switch to the general purpose
* {@link AudioRouting.OnRoutingChangedListener} class instead.
*/
@Deprecated
public void removeOnRoutingChangedListener(OnRoutingChangedListener listener) {
removeOnRoutingChangedListener((AudioRouting.OnRoutingChangedListener) listener);
}
/**
* Sends device list change notification to all listeners.
*/
private void broadcastRoutingChange() {
AudioManager.resetAudioPortGeneration();
baseUpdateDeviceId(getRoutedDevice());
synchronized (mRoutingChangeListeners) {
for (NativeRoutingEventHandlerDelegate delegate : mRoutingChangeListeners.values()) {
delegate.notifyClient();
}
}
}
//--------------------------------------------------------------------------
// Codec notifications
//--------------------
// OnCodecFormatChangedListener notifications uses an instance
// of ListenerList to manage its listeners.
private final Utils.ListenerList