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. 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 data sample rate in Hz
*/
public int getSampleRate() {
return mSampleRate;
}
/**
* Returns the current playback rate in Hz.
*/
public int getPlaybackRate() {
return native_get_playback_rate();
}
/**
* Returns the configured audio data format. See {@link AudioFormat#ENCODING_PCM_16BIT}
* and {@link AudioFormat#ENCODING_PCM_8BIT}.
*/
public int getAudioFormat() {
return mAudioFormat;
}
/**
* Returns the type of audio stream this AudioTrack is configured for.
* 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}, or {@link AudioManager#STREAM_DTMF}.
*/
public int getStreamType() {
return mStreamType;
}
/**
* Returns the configured channel configuration.
* See {@link AudioFormat#CHANNEL_OUT_MONO}
* and {@link AudioFormat#CHANNEL_OUT_STEREO}.
*/
public int getChannelConfiguration() {
return mChannelConfiguration;
}
/**
* 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_INITIALIZED
* @see #STATE_NO_STATIC_DATA
* @see #STATE_UNINITIALIZED
*/
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) {
return mPlayState;
}
}
/**
* Returns the "native frame count", derived from the bufferSizeInBytes specified at
* creation time and converted to frame units.
* If track's creation mode is {@link #MODE_STATIC},
* it is equal to the specified bufferSizeInBytes converted to frame units.
* If track's creation mode is {@link #MODE_STREAM},
* it is typically greater than or equal to the specified bufferSizeInBytes converted to frame
* units; it may be rounded up to a larger value if needed by the target device implementation.
* @deprecated Only accessible by subclasses, which are not recommended for AudioTrack.
* See {@link AudioManager#getProperty(String)} for key
* {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}.
*/
@Deprecated
protected int getNativeFrameCount() {
return native_get_native_frame_count();
}
/**
* 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 flush(), reload(), and stop().
*/
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
*/
public int getLatency() {
return native_get_latency();
}
/**
* 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 minimum buffer size required for the successful creation of an AudioTrack
* object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't
* guarantee a smooth playback under load, and higher values should be chosen according to
* the expected frequency at which the buffer will be refilled with additional data to play.
* For example, if you intend to dynamically set the source sample rate of an AudioTrack
* to a higher value than the initial source sample rate, be sure to configure the buffer size
* based on the highest planned sample rate.
* @param sampleRateInHz the source sample rate expressed in Hz.
* @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 ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) {
// input channel configuration features unsupported channels
loge("getMinBufferSize(): Invalid channel configuration.");
return ERROR_BAD_VALUE;
} else {
channelCount = Integer.bitCount(channelConfig);
}
}
if (!AudioFormat.isValidEncoding(audioFormat)) {
loge("getMinBufferSize(): Invalid audio format.");
return ERROR_BAD_VALUE;
}
// sample rate, note these values are subject to change
if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > 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 once per second until the reported timestamps
* show that the audio clock is stable.
* Thereafter, query for a new timestamp approximately once every 10 seconds to once per minute.
* Calling this method more often is inefficient.
* It is also counter-productive to call this method more often than recommended,
* because the short-term differences between successive timestamp reports are not meaningful.
* If you need a high-resolution mapping between frame position and presentation time,
* consider implementing that at application level, based on low-resolution timestamps.
*
* The audio data at the returned position may either already have been
* presented, or may have not yet been presented but is committed to be presented.
* It is not possible to request the time corresponding to a particular position,
* or to request the (fractional) position corresponding to a particular time.
* If you need such features, consider implementing them at application level.
*
* @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 if available,
* the AudioTimestamp instance is filled in with a position in frame units, together
* with the estimated time when that frame was presented or is committed to
* be presented.
* In the case that no timestamp is available, any supplied instance is left unaltered.
* A timestamp may be temporarily unavailable while the audio clock is stabilizing,
* or during and immediately 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;
}
//--------------------------------------------------------------------------
// 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 NativeEventHandlerDelegate(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.
*/
public int setStereoVolume(float leftGain, float rightGain) {
if (isRestricted()) {
return SUCCESS;
}
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
leftGain = clampGainOrLevel(leftGain);
rightGain = clampGainOrLevel(rightGain);
native_setVolume(leftGain, rightGain);
return SUCCESS;
}
/**
* 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);
}
/**
* 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)}.
* @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 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
* @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.
* 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 expressed in frames
* Zero corresponds to start of buffer.
* The position must not be greater than the buffer size in frames, or negative.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setPlaybackHeadPosition(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.
* @param loopCount the number of times the loop is looped.
* A value of -1 means infinite looping, and 0 disables looping.
* @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
* {@link #ERROR_INVALID_OPERATION}
*/
public int setLoopPoints(int startInFrames, int endInFrames, 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 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.
* If track's creation mode is {@link #MODE_STATIC}, you must have called write() prior.
*
* @throws IllegalStateException
*/
public void play()
throws IllegalStateException {
if (mState != STATE_INITIALIZED) {
throw new IllegalStateException("play() called on uninitialized AudioTrack.");
}
if (isRestricted()) {
setVolume(0);
}
synchronized(mPlayStateLock) {
native_start();
mPlayState = PLAYSTATE_PLAYING;
}
}
private boolean isRestricted() {
try {
final int usage = AudioAttributes.usageForLegacyStreamType(mStreamType);
final int mode = mAppOps.checkAudioOperation(AppOpsManager.OP_PLAY_AUDIO, usage,
Process.myUid(), ActivityThread.currentPackageName());
return mode != AppOpsManager.MODE_ALLOWED;
} catch (RemoteException e) {
return 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();
mPlayState = PLAYSTATE_STOPPED;
}
}
/**
* 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.");
}
//logd("pause()");
// pause playback
synchronized(mPlayStateLock) {
native_pause();
mPlayState = PLAYSTATE_PAUSED;
}
}
//---------------------------------------------------------
// Audio data supply
//--------------------
/**
* Flushes the audio data currently queued for playback. Any data that has
* not been played back will be discarded. No-op if not stopped or paused,
* or if the track's creation mode is not {@link #MODE_STREAM}.
*/
public void flush() {
if (mState == STATE_INITIALIZED) {
// flush the data in native layer
native_flush();
}
}
/**
* Writes the audio data to the audio sink for playback (streaming mode),
* or copies audio data for later playback (static buffer mode).
* In streaming mode, will block until all data has been written to the audio sink.
* In static buffer mode, copies the data to the buffer starting at offset 0.
* Note that the actual playback of this data might occur after this function
* returns. This function is thread safe with respect to {@link #stop} calls,
* in which case all of the specified data might not be written to the audio sink.
*
* @param audioData the array that holds the data to play.
* @param offsetInBytes the offset expressed in bytes in audioData where the data to play
* starts.
* @param sizeInBytes the number of bytes to read in audioData after the offset.
* @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION}
* if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if
* the parameters don't resolve to valid data and indexes, or
* {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and
* needs to be recreated.
*/
public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) {
if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes < 0) // detect integer overflow
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat,
true /*isBlocking*/);
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).
* In streaming mode, will block until all data has been written to the audio sink.
* In static buffer mode, copies the data to the buffer starting at offset 0.
* Note that the actual playback of this data might occur after this function
* returns. This function is thread safe with respect to {@link #stop} calls,
* in which case all of the specified data might not be written to the audio sink.
*
* @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.
* @param sizeInShorts the number of shorts to read in audioData after the offset.
* @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION}
* if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if
* the parameters don't resolve to valid data and indexes.
*/
public int write(short[] audioData, int offsetInShorts, int sizeInShorts) {
if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) {
return ERROR_INVALID_OPERATION;
}
if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
|| (offsetInShorts + sizeInShorts < 0) // detect integer overflow
|| (offsetInShorts + sizeInShorts > audioData.length)) {
return ERROR_BAD_VALUE;
}
int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat);
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).
* In static buffer mode, copies the data to the buffer starting at offset 0,
* and the write mode is ignored.
* In streaming mode, the blocking behavior will depend on the write mode.
*
* Note that the actual playback of this data might occur after this function
* returns. This function is thread safe with respect to {@link #stop} calls,
* in which case all of the specified data might not be written to the audio sink.
*
* @param audioData the array that holds the data to play.
* The implementation does not clip for sample values within the nominal range
* [-1.0f, 1.0f], provided that all gains in the audio pipeline are
* less than or equal to unity (1.0f), and in the absence of post-processing effects
* that could add energy, such as reverb. For the convenience of applications
* that compute samples using filters with non-unity gain,
* sample values +3 dB beyond the nominal range are permitted.
* However such values may eventually be limited or clipped, depending on various gains
* and later processing in the audio path. Therefore applications are encouraged
* to provide samples values within the nominal range.
* @param offsetInFloats the offset, expressed as a number of floats,
* in audioData where the data to play starts.
* @param sizeInFloats the number of floats to read in audioData after the offset.
* @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no
* effect in static mode.
* 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(float level) {
if (isRestricted()) {
return SUCCESS;
}
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
level = clampGainOrLevel(level);
int err = native_setAuxEffectSendLevel(level);
return err == 0 ? SUCCESS : ERROR;
}
//---------------------------------------------------------
// Interface definitions
//--------------------
/**
* Interface definition for a callback to be invoked when the playback head position of
* an AudioTrack has reached a notification marker or has increased by a certain period.
*/
public interface OnPlaybackPositionUpdateListener {
/**
* Called on the listener to notify it that the previously set marker has been reached
* by the playback head.
*/
void onMarkerReached(AudioTrack track);
/**
* Called on the listener to periodically notify it that the playback head has reached
* a multiple of the notification period.
*/
void onPeriodicNotification(AudioTrack track);
}
//---------------------------------------------------------
// Inner classes
//--------------------
/**
* Helper class to handle the forwarding of native events to the appropriate listener
* (potentially) handled in a different thread
*/
private class NativeEventHandlerDelegate {
private final Handler mHandler;
NativeEventHandlerDelegate(final AudioTrack track,
final OnPlaybackPositionUpdateListener listener,
Handler handler) {
// find the looper for our new event handler
Looper looper;
if (handler != null) {
looper = handler.getLooper();
} else {
// no given handler, use the looper the AudioTrack was created in
looper = mInitializationLooper;
}
// construct the event handler with this looper
if (looper != null) {
// implement the event handler delegate
mHandler = new Handler(looper) {
@Override
public void handleMessage(Message msg) {
if (track == null) {
return;
}
switch(msg.what) {
case NATIVE_EVENT_MARKER:
if (listener != null) {
listener.onMarkerReached(track);
}
break;
case NATIVE_EVENT_NEW_POS:
if (listener != null) {
listener.onPeriodicNotification(track);
}
break;
default:
loge("Unknown native event type: " + msg.what);
break;
}
}
};
} else {
mHandler = null;
}
}
Handler getHandler() {
return mHandler;
}
}
//---------------------------------------------------------
// Java methods called from the native side
//--------------------
@SuppressWarnings("unused")
private static void postEventFromNative(Object audiotrack_ref,
int what, int arg1, int arg2, Object obj) {
//logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2);
AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get();
if (track == null) {
return;
}
NativeEventHandlerDelegate delegate = track.mEventHandlerDelegate;
if (delegate != null) {
Handler handler = delegate.getHandler();
if (handler != null) {
Message m = handler.obtainMessage(what, arg1, arg2, obj);
handler.sendMessage(m);
}
}
}
//---------------------------------------------------------
// Native methods called from the Java side
//--------------------
// post-condition: mStreamType is overwritten with a value
// that reflects the audio attributes (e.g. an AudioAttributes object with a usage of
// AudioAttributes.USAGE_MEDIA will map to AudioManager.STREAM_MUSIC
private native final int native_setup(Object /*WeakReference
* 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
{
//---------------------------------------------------------
// 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;
/** Minimum value for sample rate */
private static final int SAMPLE_RATE_HZ_MIN = 4000;
/** Maximum value for sample rate */
private static final int SAMPLE_RATE_HZ_MAX = 96000;
/** Maximum value for AudioTrack channel count */
private static final int CHANNEL_COUNT_MAX = 8;
/** 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
// 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;
/**
* 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;
// 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
/**
* 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;
private final static String TAG = "android.media.AudioTrack";
/** @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 in {@link #write(ByteBuffer, int, int)}
*/
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 in
* {@link #write(ByteBuffer, int, int)}.
*/
public final static int WRITE_NON_BLOCKING = 1;
//--------------------------------------------------------------------------
// Member variables
//--------------------
/**
* Indicates the state of the AudioTrack instance.
*/
private int mState = STATE_UNINITIALIZED;
/**
* Indicates the play state of the AudioTrack instance.
*/
private int mPlayState = PLAYSTATE_STOPPED;
/**
* Lock to make sure mPlayState updates are reflecting the actual state of the object.
*/
private final Object mPlayStateLock = new Object();
/**
* Sizes of the native audio buffer.
*/
private int mNativeBufferSizeInBytes = 0;
private int mNativeBufferSizeInFrames = 0;
/**
* Handler for events coming from the native code.
*/
private NativeEventHandlerDelegate mEventHandlerDelegate;
/**
* Looper associated with the thread that creates the AudioTrack instance.
*/
private final Looper mInitializationLooper;
/**
* The audio data source sampling rate in Hz.
*/
private int mSampleRate; // initialized by all constructors
/**
* The number of audio output channels (1 is mono, 2 is stereo).
*/
private int mChannelCount = 1;
/**
* The audio channel mask.
*/
private int mChannels = 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}.
*/
private int mStreamType = AudioManager.STREAM_MUSIC;
private final AudioAttributes mAttributes;
/**
* The way audio is consumed by the audio sink, streaming or static.
*/
private int mDataLoadMode = MODE_STREAM;
/**
* The current audio channel configuration.
*/
private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO;
/**
* The encoding of the audio samples.
* @see AudioFormat#ENCODING_PCM_8BIT
* @see AudioFormat#ENCODING_PCM_16BIT
* @see AudioFormat#ENCODING_PCM_FLOAT
*/
private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT;
/**
* Audio session ID
*/
private int mSessionId = AudioSystem.AUDIO_SESSION_ALLOCATE;
/**
* Reference to the app-ops service.
*/
private final IAppOpsService mAppOps;
//--------------------------------
// Used exclusively by native code
//--------------------
/**
* Accessed by native methods: provides access to C++ AudioTrack object.
*/
@SuppressWarnings("unused")
private 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")
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.
* @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.
* If track's creation mode is {@link #MODE_STREAM}, you can write data into
* this buffer in chunks less than or equal to this size, and it is typical to use
* chunks of 1/2 of the total size to permit double-buffering.
* If the track's creation mode is {@link #MODE_STATIC},
* this is the maximum length sample, or audio clip, that can be played by this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}
* @throws java.lang.IllegalArgumentException
*/
public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode)
throws IllegalArgumentException {
this(streamType, sampleRateInHz, channelConfig, audioFormat,
bufferSizeInBytes, mode, AudioSystem.AUDIO_SESSION_ALLOCATE);
}
/**
* 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.
* @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 buffer where audio data is read
* from for playback. If using the AudioTrack in streaming mode, you can write data into
* this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
* this is the maximum size of the sound that will be played for this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @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
*/
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);
}
/**
* 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 buffer where audio data is read
* from for playback. If using the AudioTrack in streaming mode, you can write data into
* this buffer in smaller chunks than this size. If using the AudioTrack in static mode,
* this is the maximum size of the sound that will be played for this instance.
* See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size
* for the successful creation of an AudioTrack instance in streaming mode. Using values
* smaller than getMinBufferSize() will result in an initialization failure.
* @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 {
// mState already == STATE_UNINITIALIZED
if (attributes == null) {
throw new IllegalArgumentException("Illegal null AudioAttributes");
}
if (format == null) {
throw new IllegalArgumentException("Illegal null AudioFormat");
}
// remember which looper is associated with the AudioTrack instantiation
Looper looper;
if ((looper = Looper.myLooper()) == null) {
looper = Looper.getMainLooper();
}
int rate = 0;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE) != 0)
{
rate = format.getSampleRate();
} else {
rate = AudioSystem.getPrimaryOutputSamplingRate();
if (rate <= 0) {
rate = 44100;
}
}
int channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0)
{
channelMask = format.getChannelMask();
}
int encoding = AudioFormat.ENCODING_DEFAULT;
if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
encoding = format.getEncoding();
}
audioParamCheck(rate, channelMask, encoding, mode);
mStreamType = AudioSystem.STREAM_DEFAULT;
audioBuffSizeCheck(bufferSizeInBytes);
mInitializationLooper = looper;
IBinder b = ServiceManager.getService(Context.APP_OPS_SERVICE);
mAppOps = IAppOpsService.Stub.asInterface(b);
mAttributes = (new AudioAttributes.Builder(attributes).build());
if (sessionId < 0) {
throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
}
int[] session = new int[1];
session[0] = sessionId;
// native initialization
int initResult = native_setup(new WeakReference
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 the number of floats that were written, or {@link #ERROR_INVALID_OPERATION}
* if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if
* the parameters don't resolve to valid data and indexes.
*/
public int write(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;
}
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).
* In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write
* mode is ignored.
* In streaming mode, the blocking behavior will depend on the write mode.
* @param audioData the buffer that holds the data to play, starting at the position reported
* by audioData.position().
*
Note that upon return, the buffer position (audioData.position()) will
* have been advanced to reflect the amount of data that was successfully written to
* the AudioTrack.
* @param sizeInBytes number of bytes to write.
*
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 0 or a positive number of bytes that were written, or
* {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}
*/
public int write(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;
}
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;
}
/**
* Notifies the native resource to reuse the audio data already loaded in the native
* layer, that is to rewind to start of buffer.
* The track's creation mode must be {@link #MODE_STATIC}.
* @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();
}
//--------------------------------------------------------------------------
// 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)}.
*