AudioTrack.java revision 30d794360f35592554403922bcc07835fea4737b
1/* 2 * Copyright (C) 2008 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17package android.media; 18 19import java.lang.annotation.Retention; 20import java.lang.annotation.RetentionPolicy; 21import java.lang.ref.WeakReference; 22import java.lang.Math; 23import java.nio.ByteBuffer; 24import java.nio.ByteOrder; 25import java.nio.NioUtils; 26import java.util.Collection; 27 28import android.annotation.IntDef; 29import android.annotation.NonNull; 30import android.annotation.SystemApi; 31import android.app.ActivityThread; 32import android.app.AppOpsManager; 33import android.content.Context; 34import android.os.Handler; 35import android.os.IBinder; 36import android.os.Looper; 37import android.os.Message; 38import android.os.Process; 39import android.os.RemoteException; 40import android.os.ServiceManager; 41import android.util.ArrayMap; 42import android.util.Log; 43 44import com.android.internal.app.IAppOpsService; 45 46 47/** 48 * The AudioTrack class manages and plays a single audio resource for Java applications. 49 * It allows streaming of PCM audio buffers to the audio sink for playback. This is 50 * achieved by "pushing" the data to the AudioTrack object using one of the 51 * {@link #write(byte[], int, int)}, {@link #write(short[], int, int)}, 52 * and {@link #write(float[], int, int, int)} methods. 53 * 54 * <p>An AudioTrack instance can operate under two modes: static or streaming.<br> 55 * In Streaming mode, the application writes a continuous stream of data to the AudioTrack, using 56 * one of the {@code write()} methods. These are blocking and return when the data has been 57 * transferred from the Java layer to the native layer and queued for playback. The streaming 58 * mode is most useful when playing blocks of audio data that for instance are: 59 * 60 * <ul> 61 * <li>too big to fit in memory because of the duration of the sound to play,</li> 62 * <li>too big to fit in memory because of the characteristics of the audio data 63 * (high sampling rate, bits per sample ...)</li> 64 * <li>received or generated while previously queued audio is playing.</li> 65 * </ul> 66 * 67 * The static mode should be chosen when dealing with short sounds that fit in memory and 68 * that need to be played with the smallest latency possible. The static mode will 69 * therefore be preferred for UI and game sounds that are played often, and with the 70 * smallest overhead possible. 71 * 72 * <p>Upon creation, an AudioTrack object initializes its associated audio buffer. 73 * The size of this buffer, specified during the construction, determines how long an AudioTrack 74 * can play before running out of data.<br> 75 * For an AudioTrack using the static mode, this size is the maximum size of the sound that can 76 * be played from it.<br> 77 * For the streaming mode, data will be written to the audio sink in chunks of 78 * sizes less than or equal to the total buffer size. 79 * 80 * AudioTrack is not final and thus permits subclasses, but such use is not recommended. 81 */ 82public class AudioTrack 83{ 84 //--------------------------------------------------------- 85 // Constants 86 //-------------------- 87 /** Minimum value for a linear gain or auxiliary effect level. 88 * This value must be exactly equal to 0.0f; do not change it. 89 */ 90 private static final float GAIN_MIN = 0.0f; 91 /** Maximum value for a linear gain or auxiliary effect level. 92 * This value must be greater than or equal to 1.0f. 93 */ 94 private static final float GAIN_MAX = 1.0f; 95 96 /** Minimum value for sample rate */ 97 private static final int SAMPLE_RATE_HZ_MIN = 4000; 98 /** Maximum value for sample rate */ 99 private static final int SAMPLE_RATE_HZ_MAX = 96000; 100 101 /** Maximum value for AudioTrack channel count */ 102 private static final int CHANNEL_COUNT_MAX = 8; 103 104 /** indicates AudioTrack state is stopped */ 105 public static final int PLAYSTATE_STOPPED = 1; // matches SL_PLAYSTATE_STOPPED 106 /** indicates AudioTrack state is paused */ 107 public static final int PLAYSTATE_PAUSED = 2; // matches SL_PLAYSTATE_PAUSED 108 /** indicates AudioTrack state is playing */ 109 public static final int PLAYSTATE_PLAYING = 3; // matches SL_PLAYSTATE_PLAYING 110 111 // keep these values in sync with android_media_AudioTrack.cpp 112 /** 113 * Creation mode where audio data is transferred from Java to the native layer 114 * only once before the audio starts playing. 115 */ 116 public static final int MODE_STATIC = 0; 117 /** 118 * Creation mode where audio data is streamed from Java to the native layer 119 * as the audio is playing. 120 */ 121 public static final int MODE_STREAM = 1; 122 123 /** @hide */ 124 @IntDef({ 125 MODE_STATIC, 126 MODE_STREAM 127 }) 128 @Retention(RetentionPolicy.SOURCE) 129 public @interface TransferMode {} 130 131 /** 132 * State of an AudioTrack that was not successfully initialized upon creation. 133 */ 134 public static final int STATE_UNINITIALIZED = 0; 135 /** 136 * State of an AudioTrack that is ready to be used. 137 */ 138 public static final int STATE_INITIALIZED = 1; 139 /** 140 * State of a successfully initialized AudioTrack that uses static data, 141 * but that hasn't received that data yet. 142 */ 143 public static final int STATE_NO_STATIC_DATA = 2; 144 145 /** 146 * Denotes a successful operation. 147 */ 148 public static final int SUCCESS = AudioSystem.SUCCESS; 149 /** 150 * Denotes a generic operation failure. 151 */ 152 public static final int ERROR = AudioSystem.ERROR; 153 /** 154 * Denotes a failure due to the use of an invalid value. 155 */ 156 public static final int ERROR_BAD_VALUE = AudioSystem.BAD_VALUE; 157 /** 158 * Denotes a failure due to the improper use of a method. 159 */ 160 public static final int ERROR_INVALID_OPERATION = AudioSystem.INVALID_OPERATION; 161 162 // Error codes: 163 // to keep in sync with frameworks/base/core/jni/android_media_AudioTrack.cpp 164 private static final int ERROR_NATIVESETUP_AUDIOSYSTEM = -16; 165 private static final int ERROR_NATIVESETUP_INVALIDCHANNELMASK = -17; 166 private static final int ERROR_NATIVESETUP_INVALIDFORMAT = -18; 167 private static final int ERROR_NATIVESETUP_INVALIDSTREAMTYPE = -19; 168 private static final int ERROR_NATIVESETUP_NATIVEINITFAILED = -20; 169 170 // Events: 171 // to keep in sync with frameworks/av/include/media/AudioTrack.h 172 /** 173 * Event id denotes when playback head has reached a previously set marker. 174 */ 175 private static final int NATIVE_EVENT_MARKER = 3; 176 /** 177 * Event id denotes when previously set update period has elapsed during playback. 178 */ 179 private static final int NATIVE_EVENT_NEW_POS = 4; 180 181 /** 182 * Event id denotes when the routing changes. 183 */ 184 private final static int NATIVE_EVENT_ROUTING_CHANGE = 1000; 185 186 187 private final static String TAG = "android.media.AudioTrack"; 188 189 190 /** @hide */ 191 @IntDef({ 192 WRITE_BLOCKING, 193 WRITE_NON_BLOCKING 194 }) 195 @Retention(RetentionPolicy.SOURCE) 196 public @interface WriteMode {} 197 198 /** 199 * The write mode indicating the write operation will block until all data has been written, 200 * to be used in {@link #write(ByteBuffer, int, int)} 201 */ 202 public final static int WRITE_BLOCKING = 0; 203 /** 204 * The write mode indicating the write operation will return immediately after 205 * queuing as much audio data for playback as possible without blocking, to be used in 206 * {@link #write(ByteBuffer, int, int)}. 207 */ 208 public final static int WRITE_NON_BLOCKING = 1; 209 210 //-------------------------------------------------------------------------- 211 // Member variables 212 //-------------------- 213 /** 214 * Indicates the state of the AudioTrack instance. 215 */ 216 private int mState = STATE_UNINITIALIZED; 217 /** 218 * Indicates the play state of the AudioTrack instance. 219 */ 220 private int mPlayState = PLAYSTATE_STOPPED; 221 /** 222 * Lock to make sure mPlayState updates are reflecting the actual state of the object. 223 */ 224 private final Object mPlayStateLock = new Object(); 225 /** 226 * Sizes of the native audio buffer. 227 * These values are set during construction and can be stale. 228 * To obtain the current native audio buffer frame count use {@link #getNativeFrameCount()}. 229 */ 230 private int mNativeBufferSizeInBytes = 0; 231 private int mNativeBufferSizeInFrames = 0; 232 /** 233 * Handler for events coming from the native code. 234 */ 235 private NativePositionEventHandlerDelegate mEventHandlerDelegate; 236 /** 237 * Looper associated with the thread that creates the AudioTrack instance. 238 */ 239 private final Looper mInitializationLooper; 240 /** 241 * The audio data source sampling rate in Hz. 242 */ 243 private int mSampleRate; // initialized by all constructors 244 /** 245 * The number of audio output channels (1 is mono, 2 is stereo). 246 */ 247 private int mChannelCount = 1; 248 /** 249 * The audio channel mask used for calling native AudioTrack 250 */ 251 private int mChannels = AudioFormat.CHANNEL_OUT_MONO; 252 253 /** 254 * The type of the audio stream to play. See 255 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 256 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 257 * {@link AudioManager#STREAM_ALARM}, {@link AudioManager#STREAM_NOTIFICATION}, and 258 * {@link AudioManager#STREAM_DTMF}. 259 */ 260 private int mStreamType = AudioManager.STREAM_MUSIC; 261 262 private final AudioAttributes mAttributes; 263 /** 264 * The way audio is consumed by the audio sink, streaming or static. 265 */ 266 private int mDataLoadMode = MODE_STREAM; 267 /** 268 * The current channel position mask, as specified on AudioTrack creation. 269 * Can be set simultaneously with channel index mask {@link #mChannelIndexMask}. 270 * May be set to {@link AudioFormat#CHANNEL_INVALID} if a channel index mask is specified. 271 */ 272 private int mChannelConfiguration = AudioFormat.CHANNEL_OUT_MONO; 273 /** 274 * The current audio channel index configuration (if specified). 275 */ 276 private int mChannelIndexMask = 0; 277 /** 278 * The encoding of the audio samples. 279 * @see AudioFormat#ENCODING_PCM_8BIT 280 * @see AudioFormat#ENCODING_PCM_16BIT 281 * @see AudioFormat#ENCODING_PCM_FLOAT 282 */ 283 private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT; 284 /** 285 * Audio session ID 286 */ 287 private int mSessionId = AudioSystem.AUDIO_SESSION_ALLOCATE; 288 /** 289 * Reference to the app-ops service. 290 */ 291 private final IAppOpsService mAppOps; 292 /** 293 * HW_AV_SYNC track AV Sync Header 294 */ 295 private ByteBuffer mAvSyncHeader = null; 296 /** 297 * HW_AV_SYNC track audio data bytes remaining to write after current AV sync header 298 */ 299 private int mAvSyncBytesRemaining = 0; 300 301 //-------------------------------- 302 // Used exclusively by native code 303 //-------------------- 304 /** 305 * Accessed by native methods: provides access to C++ AudioTrack object. 306 */ 307 @SuppressWarnings("unused") 308 private long mNativeTrackInJavaObj; 309 /** 310 * Accessed by native methods: provides access to the JNI data (i.e. resources used by 311 * the native AudioTrack object, but not stored in it). 312 */ 313 @SuppressWarnings("unused") 314 private long mJniData; 315 316 317 //-------------------------------------------------------------------------- 318 // Constructor, Finalize 319 //-------------------- 320 /** 321 * Class constructor. 322 * @param streamType the type of the audio stream. See 323 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 324 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 325 * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. 326 * @param sampleRateInHz the initial source sample rate expressed in Hz. 327 * @param channelConfig describes the configuration of the audio channels. 328 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 329 * {@link AudioFormat#CHANNEL_OUT_STEREO} 330 * @param audioFormat the format in which the audio data is represented. 331 * See {@link AudioFormat#ENCODING_PCM_16BIT}, 332 * {@link AudioFormat#ENCODING_PCM_8BIT}, 333 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 334 * @param bufferSizeInBytes the total size (in bytes) of the internal buffer where audio data is 335 * read from for playback. This should be a multiple of the frame size in bytes. 336 * <p> If the track's creation mode is {@link #MODE_STATIC}, 337 * this is the maximum length sample, or audio clip, that can be played by this instance. 338 * <p> If the track's creation mode is {@link #MODE_STREAM}, 339 * this should be the desired buffer size 340 * for the <code>AudioTrack</code> to satisfy the application's 341 * natural latency requirements. 342 * If <code>bufferSizeInBytes</code> is less than the 343 * minimum buffer size for the output sink, it is automatically increased to the minimum 344 * buffer size. 345 * The method {@link #getNativeFrameCount()} returns the 346 * actual size in frames of the native buffer created, which 347 * determines the frequency to write 348 * to the streaming <code>AudioTrack</code> to avoid underrun. 349 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 350 * @throws java.lang.IllegalArgumentException 351 */ 352 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 353 int bufferSizeInBytes, int mode) 354 throws IllegalArgumentException { 355 this(streamType, sampleRateInHz, channelConfig, audioFormat, 356 bufferSizeInBytes, mode, AudioSystem.AUDIO_SESSION_ALLOCATE); 357 } 358 359 /** 360 * Class constructor with audio session. Use this constructor when the AudioTrack must be 361 * attached to a particular audio session. The primary use of the audio session ID is to 362 * associate audio effects to a particular instance of AudioTrack: if an audio session ID 363 * is provided when creating an AudioEffect, this effect will be applied only to audio tracks 364 * and media players in the same session and not to the output mix. 365 * When an AudioTrack is created without specifying a session, it will create its own session 366 * which can be retrieved by calling the {@link #getAudioSessionId()} method. 367 * If a non-zero session ID is provided, this AudioTrack will share effects attached to this 368 * session 369 * with all other media players or audio tracks in the same session, otherwise a new session 370 * will be created for this track if none is supplied. 371 * @param streamType the type of the audio stream. See 372 * {@link AudioManager#STREAM_VOICE_CALL}, {@link AudioManager#STREAM_SYSTEM}, 373 * {@link AudioManager#STREAM_RING}, {@link AudioManager#STREAM_MUSIC}, 374 * {@link AudioManager#STREAM_ALARM}, and {@link AudioManager#STREAM_NOTIFICATION}. 375 * @param sampleRateInHz the initial source sample rate expressed in Hz. 376 * @param channelConfig describes the configuration of the audio channels. 377 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 378 * {@link AudioFormat#CHANNEL_OUT_STEREO} 379 * @param audioFormat the format in which the audio data is represented. 380 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 381 * {@link AudioFormat#ENCODING_PCM_8BIT}, 382 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 383 * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read 384 * from for playback. If using the AudioTrack in streaming mode, you can write data into 385 * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, 386 * this is the maximum size of the sound that will be played for this instance. 387 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 388 * for the successful creation of an AudioTrack instance in streaming mode. Using values 389 * smaller than getMinBufferSize() will result in an initialization failure. 390 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM} 391 * @param sessionId Id of audio session the AudioTrack must be attached to 392 * @throws java.lang.IllegalArgumentException 393 */ 394 public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat, 395 int bufferSizeInBytes, int mode, int sessionId) 396 throws IllegalArgumentException { 397 // mState already == STATE_UNINITIALIZED 398 this((new AudioAttributes.Builder()) 399 .setLegacyStreamType(streamType) 400 .build(), 401 (new AudioFormat.Builder()) 402 .setChannelMask(channelConfig) 403 .setEncoding(audioFormat) 404 .setSampleRate(sampleRateInHz) 405 .build(), 406 bufferSizeInBytes, 407 mode, sessionId); 408 } 409 410 /** 411 * Class constructor with {@link AudioAttributes} and {@link AudioFormat}. 412 * @param attributes a non-null {@link AudioAttributes} instance. 413 * @param format a non-null {@link AudioFormat} instance describing the format of the data 414 * that will be played through this AudioTrack. See {@link AudioFormat.Builder} for 415 * configuring the audio format parameters such as encoding, channel mask and sample rate. 416 * @param bufferSizeInBytes the total size (in bytes) of the buffer where audio data is read 417 * from for playback. If using the AudioTrack in streaming mode, you can write data into 418 * this buffer in smaller chunks than this size. If using the AudioTrack in static mode, 419 * this is the maximum size of the sound that will be played for this instance. 420 * See {@link #getMinBufferSize(int, int, int)} to determine the minimum required buffer size 421 * for the successful creation of an AudioTrack instance in streaming mode. Using values 422 * smaller than getMinBufferSize() will result in an initialization failure. 423 * @param mode streaming or static buffer. See {@link #MODE_STATIC} and {@link #MODE_STREAM}. 424 * @param sessionId ID of audio session the AudioTrack must be attached to, or 425 * {@link AudioManager#AUDIO_SESSION_ID_GENERATE} if the session isn't known at construction 426 * time. See also {@link AudioManager#generateAudioSessionId()} to obtain a session ID before 427 * construction. 428 * @throws IllegalArgumentException 429 */ 430 public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes, 431 int mode, int sessionId) 432 throws IllegalArgumentException { 433 // mState already == STATE_UNINITIALIZED 434 435 if (attributes == null) { 436 throw new IllegalArgumentException("Illegal null AudioAttributes"); 437 } 438 if (format == null) { 439 throw new IllegalArgumentException("Illegal null AudioFormat"); 440 } 441 442 // remember which looper is associated with the AudioTrack instantiation 443 Looper looper; 444 if ((looper = Looper.myLooper()) == null) { 445 looper = Looper.getMainLooper(); 446 } 447 448 int rate = 0; 449 if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_SAMPLE_RATE) != 0) 450 { 451 rate = format.getSampleRate(); 452 } else { 453 rate = AudioSystem.getPrimaryOutputSamplingRate(); 454 if (rate <= 0) { 455 rate = 44100; 456 } 457 } 458 int channelIndexMask = 0; 459 if ((format.getPropertySetMask() 460 & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) { 461 channelIndexMask = format.getChannelIndexMask(); 462 } 463 int channelMask = 0; 464 if ((format.getPropertySetMask() 465 & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) { 466 channelMask = format.getChannelMask(); 467 } else if (channelIndexMask == 0) { // if no masks at all, use stereo 468 channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT 469 | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; 470 } 471 int encoding = AudioFormat.ENCODING_DEFAULT; 472 if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) { 473 encoding = format.getEncoding(); 474 } 475 audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode); 476 mStreamType = AudioSystem.STREAM_DEFAULT; 477 478 audioBuffSizeCheck(bufferSizeInBytes); 479 480 mInitializationLooper = looper; 481 IBinder b = ServiceManager.getService(Context.APP_OPS_SERVICE); 482 mAppOps = IAppOpsService.Stub.asInterface(b); 483 484 mAttributes = (new AudioAttributes.Builder(attributes).build()); 485 486 if (sessionId < 0) { 487 throw new IllegalArgumentException("Invalid audio session ID: "+sessionId); 488 } 489 490 int[] session = new int[1]; 491 session[0] = sessionId; 492 // native initialization 493 int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes, 494 mSampleRate, mChannels, mChannelIndexMask, mAudioFormat, 495 mNativeBufferSizeInBytes, mDataLoadMode, session); 496 if (initResult != SUCCESS) { 497 loge("Error code "+initResult+" when initializing AudioTrack."); 498 return; // with mState == STATE_UNINITIALIZED 499 } 500 501 mSessionId = session[0]; 502 503 if (mDataLoadMode == MODE_STATIC) { 504 mState = STATE_NO_STATIC_DATA; 505 } else { 506 mState = STATE_INITIALIZED; 507 } 508 } 509 510 /** 511 * Builder class for {@link AudioTrack} objects. 512 * Use this class to configure and create an <code>AudioTrack</code> instance. By setting audio 513 * attributes and audio format parameters, you indicate which of those vary from the default 514 * behavior on the device. 515 * <p> Here is an example where <code>Builder</code> is used to specify all {@link AudioFormat} 516 * parameters, to be used by a new <code>AudioTrack</code> instance: 517 * 518 * <pre class="prettyprint"> 519 * AudioTrack player = new AudioTrack.Builder() 520 * .setAudioAttributes(new AudioAttributes.Builder() 521 * .setUsage(AudioAttributes.USAGE_ALARM) 522 * .setContentType(CONTENT_TYPE_MUSIC) 523 * .build()) 524 * .setAudioFormat(new AudioFormat.Builder() 525 * .setEncoding(AudioFormat.ENCODING_PCM_16BIT) 526 * .setSampleRate(441000) 527 * .setChannelMask(AudioFormat.CHANNEL_OUT_STEREO) 528 * .build()) 529 * .setBufferSize(minBuffSize) 530 * .build(); 531 * </pre> 532 * <p> 533 * If the audio attributes are not set with {@link #setAudioAttributes(AudioAttributes)}, 534 * attributes comprising {@link AudioAttributes#USAGE_MEDIA} will be used. 535 * <br>If the audio format is not specified or is incomplete, its sample rate will be the 536 * default output sample rate of the device (see 537 * {@link AudioManager#PROPERTY_OUTPUT_SAMPLE_RATE}), its channel configuration will be 538 * {@link AudioFormat#CHANNEL_OUT_STEREO} and the encoding will be 539 * {@link AudioFormat#ENCODING_PCM_16BIT}. 540 * <br>If the buffer size is not specified with {@link #setBufferSizeInBytes(int)}, 541 * and the mode is {@link AudioTrack#MODE_STREAM}, the minimum buffer size is used. 542 * <br>If the transfer mode is not specified with {@link #setTransferMode(int)}, 543 * <code>MODE_STREAM</code> will be used. 544 * <br>If the session ID is not specified with {@link #setSessionId(int)}, a new one will 545 * be generated. 546 */ 547 public static class Builder { 548 private AudioAttributes mAttributes; 549 private AudioFormat mFormat; 550 private int mBufferSizeInBytes; 551 private int mSessionId = AudioManager.AUDIO_SESSION_ID_GENERATE; 552 private int mMode = MODE_STREAM; 553 554 /** 555 * Constructs a new Builder with the default values as described above. 556 */ 557 public Builder() { 558 } 559 560 /** 561 * Sets the {@link AudioAttributes}. 562 * @param attributes a non-null {@link AudioAttributes} instance that describes the audio 563 * data to be played. 564 * @return the same Builder instance. 565 * @throws IllegalArgumentException 566 */ 567 public @NonNull Builder setAudioAttributes(@NonNull AudioAttributes attributes) 568 throws IllegalArgumentException { 569 if (attributes == null) { 570 throw new IllegalArgumentException("Illegal null AudioAttributes argument"); 571 } 572 // keep reference, we only copy the data when building 573 mAttributes = attributes; 574 return this; 575 } 576 577 /** 578 * Sets the format of the audio data to be played by the {@link AudioTrack}. 579 * See {@link AudioFormat.Builder} for configuring the audio format parameters such 580 * as encoding, channel mask and sample rate. 581 * @param format a non-null {@link AudioFormat} instance. 582 * @return the same Builder instance. 583 * @throws IllegalArgumentException 584 */ 585 public @NonNull Builder setAudioFormat(@NonNull AudioFormat format) 586 throws IllegalArgumentException { 587 if (format == null) { 588 throw new IllegalArgumentException("Illegal null AudioFormat argument"); 589 } 590 // keep reference, we only copy the data when building 591 mFormat = format; 592 return this; 593 } 594 595 /** 596 * Sets the total size (in bytes) of the buffer where audio data is read from for playback. 597 * If using the {@link AudioTrack} in streaming mode 598 * (see {@link AudioTrack#MODE_STREAM}, you can write data into this buffer in smaller 599 * chunks than this size. See {@link #getMinBufferSize(int, int, int)} to determine 600 * the minimum required buffer size for the successful creation of an AudioTrack instance 601 * in streaming mode. Using values smaller than <code>getMinBufferSize()</code> will result 602 * in an exception when trying to build the <code>AudioTrack</code>. 603 * <br>If using the <code>AudioTrack</code> in static mode (see 604 * {@link AudioTrack#MODE_STATIC}), this is the maximum size of the sound that will be 605 * played by this instance. 606 * @param bufferSizeInBytes 607 * @return the same Builder instance. 608 * @throws IllegalArgumentException 609 */ 610 public @NonNull Builder setBufferSizeInBytes(int bufferSizeInBytes) 611 throws IllegalArgumentException { 612 if (bufferSizeInBytes <= 0) { 613 throw new IllegalArgumentException("Invalid buffer size " + bufferSizeInBytes); 614 } 615 mBufferSizeInBytes = bufferSizeInBytes; 616 return this; 617 } 618 619 /** 620 * Sets the mode under which buffers of audio data are transferred from the 621 * {@link AudioTrack} to the framework. 622 * @param mode one of {@link AudioTrack#MODE_STREAM}, {@link AudioTrack#MODE_STATIC}. 623 * @return the same Builder instance. 624 * @throws IllegalArgumentException 625 */ 626 public @NonNull Builder setTransferMode(@TransferMode int mode) 627 throws IllegalArgumentException { 628 switch(mode) { 629 case MODE_STREAM: 630 case MODE_STATIC: 631 mMode = mode; 632 break; 633 default: 634 throw new IllegalArgumentException("Invalid transfer mode " + mode); 635 } 636 return this; 637 } 638 639 /** 640 * Sets the session ID the {@link AudioTrack} will be attached to. 641 * @param sessionId a strictly positive ID number retrieved from another 642 * <code>AudioTrack</code> via {@link AudioTrack#getAudioSessionId()} or allocated by 643 * {@link AudioManager} via {@link AudioManager#generateAudioSessionId()}, or 644 * {@link AudioManager#AUDIO_SESSION_ID_GENERATE}. 645 * @return the same Builder instance. 646 * @throws IllegalArgumentException 647 */ 648 public @NonNull Builder setSessionId(int sessionId) 649 throws IllegalArgumentException { 650 if ((sessionId != AudioManager.AUDIO_SESSION_ID_GENERATE) && (sessionId < 1)) { 651 throw new IllegalArgumentException("Invalid audio session ID " + sessionId); 652 } 653 mSessionId = sessionId; 654 return this; 655 } 656 657 /** 658 * Builds an {@link AudioTrack} instance initialized with all the parameters set 659 * on this <code>Builder</code>. 660 * @return a new {@link AudioTrack} instance. 661 * @throws UnsupportedOperationException if the parameters set on the <code>Builder</code> 662 * were incompatible, or if they are not supported by the device. 663 */ 664 public @NonNull AudioTrack build() throws UnsupportedOperationException { 665 if (mAttributes == null) { 666 mAttributes = new AudioAttributes.Builder() 667 .setUsage(AudioAttributes.USAGE_MEDIA) 668 .build(); 669 } 670 if (mFormat == null) { 671 mFormat = new AudioFormat.Builder() 672 .setChannelMask(AudioFormat.CHANNEL_OUT_STEREO) 673 .setSampleRate(AudioSystem.getPrimaryOutputSamplingRate()) 674 .setEncoding(AudioFormat.ENCODING_DEFAULT) 675 .build(); 676 } 677 try { 678 // If the buffer size is not specified in streaming mode, 679 // use a single frame for the buffer size and let the 680 // native code figure out the minimum buffer size. 681 if (mMode == MODE_STREAM && mBufferSizeInBytes == 0) { 682 mBufferSizeInBytes = mFormat.getChannelCount() 683 * mFormat.getBytesPerSample(mFormat.getEncoding()); 684 } 685 return new AudioTrack(mAttributes, mFormat, mBufferSizeInBytes, mMode, mSessionId); 686 } catch (IllegalArgumentException e) { 687 throw new UnsupportedOperationException(e.getMessage()); 688 } 689 } 690 } 691 692 // mask of all the channels supported by this implementation 693 private static final int SUPPORTED_OUT_CHANNELS = 694 AudioFormat.CHANNEL_OUT_FRONT_LEFT | 695 AudioFormat.CHANNEL_OUT_FRONT_RIGHT | 696 AudioFormat.CHANNEL_OUT_FRONT_CENTER | 697 AudioFormat.CHANNEL_OUT_LOW_FREQUENCY | 698 AudioFormat.CHANNEL_OUT_BACK_LEFT | 699 AudioFormat.CHANNEL_OUT_BACK_RIGHT | 700 AudioFormat.CHANNEL_OUT_BACK_CENTER | 701 AudioFormat.CHANNEL_OUT_SIDE_LEFT | 702 AudioFormat.CHANNEL_OUT_SIDE_RIGHT; 703 704 // Convenience method for the constructor's parameter checks. 705 // This is where constructor IllegalArgumentException-s are thrown 706 // postconditions: 707 // mChannelCount is valid 708 // mChannels is valid 709 // mAudioFormat is valid 710 // mSampleRate is valid 711 // mDataLoadMode is valid 712 private void audioParamCheck(int sampleRateInHz, int channelConfig, int channelIndexMask, 713 int audioFormat, int mode) { 714 //-------------- 715 // sample rate, note these values are subject to change 716 if (sampleRateInHz < SAMPLE_RATE_HZ_MIN || sampleRateInHz > SAMPLE_RATE_HZ_MAX) { 717 throw new IllegalArgumentException(sampleRateInHz 718 + "Hz is not a supported sample rate."); 719 } 720 mSampleRate = sampleRateInHz; 721 722 //-------------- 723 // channel config 724 mChannelConfiguration = channelConfig; 725 726 switch (channelConfig) { 727 case AudioFormat.CHANNEL_OUT_DEFAULT: //AudioFormat.CHANNEL_CONFIGURATION_DEFAULT 728 case AudioFormat.CHANNEL_OUT_MONO: 729 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 730 mChannelCount = 1; 731 mChannels = AudioFormat.CHANNEL_OUT_MONO; 732 break; 733 case AudioFormat.CHANNEL_OUT_STEREO: 734 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 735 mChannelCount = 2; 736 mChannels = AudioFormat.CHANNEL_OUT_STEREO; 737 break; 738 default: 739 if (channelConfig == AudioFormat.CHANNEL_INVALID && channelIndexMask != 0) { 740 mChannelCount = 0; 741 break; // channel index configuration only 742 } 743 if (!isMultichannelConfigSupported(channelConfig)) { 744 // input channel configuration features unsupported channels 745 throw new IllegalArgumentException("Unsupported channel configuration."); 746 } 747 mChannels = channelConfig; 748 mChannelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 749 } 750 // check the channel index configuration (if present) 751 mChannelIndexMask = channelIndexMask; 752 if (mChannelIndexMask != 0) { 753 // restrictive: indexMask could allow up to AUDIO_CHANNEL_BITS_LOG2 754 final int indexMask = (1 << CHANNEL_COUNT_MAX) - 1; 755 if ((channelIndexMask & ~indexMask) != 0) { 756 throw new IllegalArgumentException("Unsupported channel index configuration " 757 + channelIndexMask); 758 } 759 int channelIndexCount = Integer.bitCount(channelIndexMask); 760 if (mChannelCount == 0) { 761 mChannelCount = channelIndexCount; 762 } else if (mChannelCount != channelIndexCount) { 763 throw new IllegalArgumentException("Channel count must match"); 764 } 765 } 766 767 //-------------- 768 // audio format 769 if (audioFormat == AudioFormat.ENCODING_DEFAULT) { 770 audioFormat = AudioFormat.ENCODING_PCM_16BIT; 771 } 772 773 if (!AudioFormat.isValidEncoding(audioFormat)) { 774 throw new IllegalArgumentException("Unsupported audio encoding."); 775 } 776 mAudioFormat = audioFormat; 777 778 //-------------- 779 // audio load mode 780 if (((mode != MODE_STREAM) && (mode != MODE_STATIC)) || 781 ((mode != MODE_STREAM) && !AudioFormat.isEncodingLinearPcm(mAudioFormat))) { 782 throw new IllegalArgumentException("Invalid mode."); 783 } 784 mDataLoadMode = mode; 785 } 786 787 /** 788 * Convenience method to check that the channel configuration (a.k.a channel mask) is supported 789 * @param channelConfig the mask to validate 790 * @return false if the AudioTrack can't be used with such a mask 791 */ 792 private static boolean isMultichannelConfigSupported(int channelConfig) { 793 // check for unsupported channels 794 if ((channelConfig & SUPPORTED_OUT_CHANNELS) != channelConfig) { 795 loge("Channel configuration features unsupported channels"); 796 return false; 797 } 798 final int channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 799 if (channelCount > CHANNEL_COUNT_MAX) { 800 loge("Channel configuration contains too many channels " + 801 channelCount + ">" + CHANNEL_COUNT_MAX); 802 return false; 803 } 804 // check for unsupported multichannel combinations: 805 // - FL/FR must be present 806 // - L/R channels must be paired (e.g. no single L channel) 807 final int frontPair = 808 AudioFormat.CHANNEL_OUT_FRONT_LEFT | AudioFormat.CHANNEL_OUT_FRONT_RIGHT; 809 if ((channelConfig & frontPair) != frontPair) { 810 loge("Front channels must be present in multichannel configurations"); 811 return false; 812 } 813 final int backPair = 814 AudioFormat.CHANNEL_OUT_BACK_LEFT | AudioFormat.CHANNEL_OUT_BACK_RIGHT; 815 if ((channelConfig & backPair) != 0) { 816 if ((channelConfig & backPair) != backPair) { 817 loge("Rear channels can't be used independently"); 818 return false; 819 } 820 } 821 final int sidePair = 822 AudioFormat.CHANNEL_OUT_SIDE_LEFT | AudioFormat.CHANNEL_OUT_SIDE_RIGHT; 823 if ((channelConfig & sidePair) != 0 824 && (channelConfig & sidePair) != sidePair) { 825 loge("Side channels can't be used independently"); 826 return false; 827 } 828 return true; 829 } 830 831 832 // Convenience method for the constructor's audio buffer size check. 833 // preconditions: 834 // mChannelCount is valid 835 // mAudioFormat is valid 836 // postcondition: 837 // mNativeBufferSizeInBytes is valid (multiple of frame size, positive) 838 private void audioBuffSizeCheck(int audioBufferSize) { 839 // NB: this section is only valid with PCM data. 840 // To update when supporting compressed formats 841 int frameSizeInBytes; 842 if (AudioFormat.isEncodingLinearPcm(mAudioFormat)) { 843 frameSizeInBytes = mChannelCount 844 * (AudioFormat.getBytesPerSample(mAudioFormat)); 845 } else { 846 frameSizeInBytes = 1; 847 } 848 if ((audioBufferSize % frameSizeInBytes != 0) || (audioBufferSize < 1)) { 849 throw new IllegalArgumentException("Invalid audio buffer size."); 850 } 851 852 mNativeBufferSizeInBytes = audioBufferSize; 853 mNativeBufferSizeInFrames = audioBufferSize / frameSizeInBytes; 854 } 855 856 857 /** 858 * Releases the native AudioTrack resources. 859 */ 860 public void release() { 861 // even though native_release() stops the native AudioTrack, we need to stop 862 // AudioTrack subclasses too. 863 try { 864 stop(); 865 } catch(IllegalStateException ise) { 866 // don't raise an exception, we're releasing the resources. 867 } 868 native_release(); 869 mState = STATE_UNINITIALIZED; 870 } 871 872 @Override 873 protected void finalize() { 874 native_finalize(); 875 } 876 877 //-------------------------------------------------------------------------- 878 // Getters 879 //-------------------- 880 /** 881 * Returns the minimum gain value, which is the constant 0.0. 882 * Gain values less than 0.0 will be clamped to 0.0. 883 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 884 * @return the minimum value, which is the constant 0.0. 885 */ 886 static public float getMinVolume() { 887 return GAIN_MIN; 888 } 889 890 /** 891 * Returns the maximum gain value, which is greater than or equal to 1.0. 892 * Gain values greater than the maximum will be clamped to the maximum. 893 * <p>The word "volume" in the API name is historical; this is actually a gain. 894 * expressed as a linear multiplier on sample values, where a maximum value of 1.0 895 * corresponds to a gain of 0 dB (sample values left unmodified). 896 * @return the maximum value, which is greater than or equal to 1.0. 897 */ 898 static public float getMaxVolume() { 899 return GAIN_MAX; 900 } 901 902 /** 903 * Returns the configured audio data sample rate in Hz 904 */ 905 public int getSampleRate() { 906 return mSampleRate; 907 } 908 909 /** 910 * Returns the current playback sample rate rate in Hz. 911 */ 912 public int getPlaybackRate() { 913 return native_get_playback_rate(); 914 } 915 916 /** 917 * Returns the current playback settings. 918 * See {@link #setPlaybackSettings(PlaybackSettings)} to set playback settings 919 * @return current {@link PlaybackSettings}. 920 * @throws IllegalStateException if track is not initialized. 921 */ 922 public @NonNull PlaybackSettings getPlaybackSettings() { 923 float[] floatArray = new float[2]; 924 int[] intArray = new int[2]; 925 native_get_playback_settings(floatArray, intArray); 926 return new PlaybackSettings() 927 .setSpeed(floatArray[0]) 928 .setPitch(floatArray[1]) 929 .setAudioFallbackMode(intArray[0]) 930 .setAudioStretchMode(intArray[1]); 931 } 932 933 /** 934 * Returns the configured audio data encoding. See {@link AudioFormat#ENCODING_PCM_8BIT}, 935 * {@link AudioFormat#ENCODING_PCM_16BIT}, and {@link AudioFormat#ENCODING_PCM_FLOAT}. 936 */ 937 public int getAudioFormat() { 938 return mAudioFormat; 939 } 940 941 /** 942 * Returns the type of audio stream this AudioTrack is configured for. 943 * Compare the result against {@link AudioManager#STREAM_VOICE_CALL}, 944 * {@link AudioManager#STREAM_SYSTEM}, {@link AudioManager#STREAM_RING}, 945 * {@link AudioManager#STREAM_MUSIC}, {@link AudioManager#STREAM_ALARM}, 946 * {@link AudioManager#STREAM_NOTIFICATION}, or {@link AudioManager#STREAM_DTMF}. 947 */ 948 public int getStreamType() { 949 return mStreamType; 950 } 951 952 /** 953 * Returns the configured channel position mask. 954 * <p> For example, refer to {@link AudioFormat#CHANNEL_OUT_MONO}, 955 * {@link AudioFormat#CHANNEL_OUT_STEREO}, {@link AudioFormat#CHANNEL_OUT_5POINT1}. 956 * This method may return {@link AudioFormat#CHANNEL_INVALID} if 957 * a channel index mask is used. Consider 958 * {@link #getFormat()} instead, to obtain an {@link AudioFormat}, 959 * which contains both the channel position mask and the channel index mask. 960 */ 961 public int getChannelConfiguration() { 962 return mChannelConfiguration; 963 } 964 965 /** 966 * Returns the configured <code>AudioTrack</code> format. 967 * @return an {@link AudioFormat} containing the 968 * <code>AudioTrack</code> parameters at the time of configuration. 969 */ 970 public @NonNull AudioFormat getFormat() { 971 AudioFormat.Builder builder = new AudioFormat.Builder() 972 .setSampleRate(mSampleRate) 973 .setEncoding(mAudioFormat); 974 if (mChannelConfiguration != AudioFormat.CHANNEL_INVALID) { 975 builder.setChannelMask(mChannelConfiguration); 976 } 977 if (mChannelIndexMask != AudioFormat.CHANNEL_INVALID /* 0 */) { 978 builder.setChannelIndexMask(mChannelIndexMask); 979 } 980 return builder.build(); 981 } 982 983 /** 984 * Returns the configured number of channels. 985 */ 986 public int getChannelCount() { 987 return mChannelCount; 988 } 989 990 /** 991 * Returns the state of the AudioTrack instance. This is useful after the 992 * AudioTrack instance has been created to check if it was initialized 993 * properly. This ensures that the appropriate resources have been acquired. 994 * @see #STATE_INITIALIZED 995 * @see #STATE_NO_STATIC_DATA 996 * @see #STATE_UNINITIALIZED 997 */ 998 public int getState() { 999 return mState; 1000 } 1001 1002 /** 1003 * Returns the playback state of the AudioTrack instance. 1004 * @see #PLAYSTATE_STOPPED 1005 * @see #PLAYSTATE_PAUSED 1006 * @see #PLAYSTATE_PLAYING 1007 */ 1008 public int getPlayState() { 1009 synchronized (mPlayStateLock) { 1010 return mPlayState; 1011 } 1012 } 1013 1014 /** 1015 * Returns the "native frame count" of the <code>AudioTrack</code> buffer. 1016 * <p> If the track's creation mode is {@link #MODE_STATIC}, 1017 * it is equal to the specified bufferSizeInBytes on construction, converted to frame units. 1018 * A static track's native frame count will not change. 1019 * <p> If the track's creation mode is {@link #MODE_STREAM}, 1020 * it is greater than or equal to the specified bufferSizeInBytes converted to frame units. 1021 * For streaming tracks, this value may be rounded up to a larger value if needed by 1022 * the target output sink, and 1023 * if the track is subsequently routed to a different output sink, the native 1024 * frame count may enlarge to accommodate. 1025 * See also {@link AudioManager#getProperty(String)} for key 1026 * {@link AudioManager#PROPERTY_OUTPUT_FRAMES_PER_BUFFER}. 1027 * @return current size in frames of the audio track buffer. 1028 * @throws IllegalStateException 1029 */ 1030 public int getNativeFrameCount() throws IllegalStateException { 1031 return native_get_native_frame_count(); 1032 } 1033 1034 /** 1035 * Returns marker position expressed in frames. 1036 * @return marker position in wrapping frame units similar to {@link #getPlaybackHeadPosition}, 1037 * or zero if marker is disabled. 1038 */ 1039 public int getNotificationMarkerPosition() { 1040 return native_get_marker_pos(); 1041 } 1042 1043 /** 1044 * Returns the notification update period expressed in frames. 1045 * Zero means that no position update notifications are being delivered. 1046 */ 1047 public int getPositionNotificationPeriod() { 1048 return native_get_pos_update_period(); 1049 } 1050 1051 /** 1052 * Returns the playback head position expressed in frames. 1053 * Though the "int" type is signed 32-bits, the value should be reinterpreted as if it is 1054 * unsigned 32-bits. That is, the next position after 0x7FFFFFFF is (int) 0x80000000. 1055 * This is a continuously advancing counter. It will wrap (overflow) periodically, 1056 * for example approximately once every 27:03:11 hours:minutes:seconds at 44.1 kHz. 1057 * It is reset to zero by {@link #flush()}, {@link #reloadStaticData()}, and {@link #stop()}. 1058 * If the track's creation mode is {@link #MODE_STATIC}, the return value indicates 1059 * the total number of frames played since reset, 1060 * <i>not</i> the current offset within the buffer. 1061 */ 1062 public int getPlaybackHeadPosition() { 1063 return native_get_position(); 1064 } 1065 1066 /** 1067 * Returns this track's estimated latency in milliseconds. This includes the latency due 1068 * to AudioTrack buffer size, AudioMixer (if any) and audio hardware driver. 1069 * 1070 * DO NOT UNHIDE. The existing approach for doing A/V sync has too many problems. We need 1071 * a better solution. 1072 * @hide 1073 */ 1074 public int getLatency() { 1075 return native_get_latency(); 1076 } 1077 1078 /** 1079 * Returns the output sample rate in Hz for the specified stream type. 1080 */ 1081 static public int getNativeOutputSampleRate(int streamType) { 1082 return native_get_output_sample_rate(streamType); 1083 } 1084 1085 /** 1086 * Returns the minimum buffer size required for the successful creation of an AudioTrack 1087 * object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't 1088 * guarantee a smooth playback under load, and higher values should be chosen according to 1089 * the expected frequency at which the buffer will be refilled with additional data to play. 1090 * For example, if you intend to dynamically set the source sample rate of an AudioTrack 1091 * to a higher value than the initial source sample rate, be sure to configure the buffer size 1092 * based on the highest planned sample rate. 1093 * @param sampleRateInHz the source sample rate expressed in Hz. 1094 * @param channelConfig describes the configuration of the audio channels. 1095 * See {@link AudioFormat#CHANNEL_OUT_MONO} and 1096 * {@link AudioFormat#CHANNEL_OUT_STEREO} 1097 * @param audioFormat the format in which the audio data is represented. 1098 * See {@link AudioFormat#ENCODING_PCM_16BIT} and 1099 * {@link AudioFormat#ENCODING_PCM_8BIT}, 1100 * and {@link AudioFormat#ENCODING_PCM_FLOAT}. 1101 * @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed, 1102 * or {@link #ERROR} if unable to query for output properties, 1103 * or the minimum buffer size expressed in bytes. 1104 */ 1105 static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) { 1106 int channelCount = 0; 1107 switch(channelConfig) { 1108 case AudioFormat.CHANNEL_OUT_MONO: 1109 case AudioFormat.CHANNEL_CONFIGURATION_MONO: 1110 channelCount = 1; 1111 break; 1112 case AudioFormat.CHANNEL_OUT_STEREO: 1113 case AudioFormat.CHANNEL_CONFIGURATION_STEREO: 1114 channelCount = 2; 1115 break; 1116 default: 1117 if (!isMultichannelConfigSupported(channelConfig)) { 1118 loge("getMinBufferSize(): Invalid channel configuration."); 1119 return ERROR_BAD_VALUE; 1120 } else { 1121 channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig); 1122 } 1123 } 1124 1125 if (!AudioFormat.isValidEncoding(audioFormat)) { 1126 loge("getMinBufferSize(): Invalid audio format."); 1127 return ERROR_BAD_VALUE; 1128 } 1129 1130 // sample rate, note these values are subject to change 1131 if ( (sampleRateInHz < SAMPLE_RATE_HZ_MIN) || (sampleRateInHz > SAMPLE_RATE_HZ_MAX) ) { 1132 loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate."); 1133 return ERROR_BAD_VALUE; 1134 } 1135 1136 int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat); 1137 if (size <= 0) { 1138 loge("getMinBufferSize(): error querying hardware"); 1139 return ERROR; 1140 } 1141 else { 1142 return size; 1143 } 1144 } 1145 1146 /** 1147 * Returns the audio session ID. 1148 * 1149 * @return the ID of the audio session this AudioTrack belongs to. 1150 */ 1151 public int getAudioSessionId() { 1152 return mSessionId; 1153 } 1154 1155 /** 1156 * Poll for a timestamp on demand. 1157 * <p> 1158 * If you need to track timestamps during initial warmup or after a routing or mode change, 1159 * you should request a new timestamp once per second until the reported timestamps 1160 * show that the audio clock is stable. 1161 * Thereafter, query for a new timestamp approximately once every 10 seconds to once per minute. 1162 * Calling this method more often is inefficient. 1163 * It is also counter-productive to call this method more often than recommended, 1164 * because the short-term differences between successive timestamp reports are not meaningful. 1165 * If you need a high-resolution mapping between frame position and presentation time, 1166 * consider implementing that at application level, based on low-resolution timestamps. 1167 * <p> 1168 * The audio data at the returned position may either already have been 1169 * presented, or may have not yet been presented but is committed to be presented. 1170 * It is not possible to request the time corresponding to a particular position, 1171 * or to request the (fractional) position corresponding to a particular time. 1172 * If you need such features, consider implementing them at application level. 1173 * 1174 * @param timestamp a reference to a non-null AudioTimestamp instance allocated 1175 * and owned by caller. 1176 * @return true if a timestamp is available, or false if no timestamp is available. 1177 * If a timestamp if available, 1178 * the AudioTimestamp instance is filled in with a position in frame units, together 1179 * with the estimated time when that frame was presented or is committed to 1180 * be presented. 1181 * In the case that no timestamp is available, any supplied instance is left unaltered. 1182 * A timestamp may be temporarily unavailable while the audio clock is stabilizing, 1183 * or during and immediately after a route change. 1184 */ 1185 // Add this text when the "on new timestamp" API is added: 1186 // Use if you need to get the most recent timestamp outside of the event callback handler. 1187 public boolean getTimestamp(AudioTimestamp timestamp) 1188 { 1189 if (timestamp == null) { 1190 throw new IllegalArgumentException(); 1191 } 1192 // It's unfortunate, but we have to either create garbage every time or use synchronized 1193 long[] longArray = new long[2]; 1194 int ret = native_get_timestamp(longArray); 1195 if (ret != SUCCESS) { 1196 return false; 1197 } 1198 timestamp.framePosition = longArray[0]; 1199 timestamp.nanoTime = longArray[1]; 1200 return true; 1201 } 1202 1203 1204 //-------------------------------------------------------------------------- 1205 // Initialization / configuration 1206 //-------------------- 1207 /** 1208 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 1209 * for each periodic playback head position update. 1210 * Notifications will be received in the same thread as the one in which the AudioTrack 1211 * instance was created. 1212 * @param listener 1213 */ 1214 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) { 1215 setPlaybackPositionUpdateListener(listener, null); 1216 } 1217 1218 /** 1219 * Sets the listener the AudioTrack notifies when a previously set marker is reached or 1220 * for each periodic playback head position update. 1221 * Use this method to receive AudioTrack events in the Handler associated with another 1222 * thread than the one in which you created the AudioTrack instance. 1223 * @param listener 1224 * @param handler the Handler that will receive the event notification messages. 1225 */ 1226 public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener, 1227 Handler handler) { 1228 if (listener != null) { 1229 mEventHandlerDelegate = new NativePositionEventHandlerDelegate(this, listener, handler); 1230 } else { 1231 mEventHandlerDelegate = null; 1232 } 1233 } 1234 1235 1236 private static float clampGainOrLevel(float gainOrLevel) { 1237 if (Float.isNaN(gainOrLevel)) { 1238 throw new IllegalArgumentException(); 1239 } 1240 if (gainOrLevel < GAIN_MIN) { 1241 gainOrLevel = GAIN_MIN; 1242 } else if (gainOrLevel > GAIN_MAX) { 1243 gainOrLevel = GAIN_MAX; 1244 } 1245 return gainOrLevel; 1246 } 1247 1248 1249 /** 1250 * Sets the specified left and right output gain values on the AudioTrack. 1251 * <p>Gain values are clamped to the closed interval [0.0, max] where 1252 * max is the value of {@link #getMaxVolume}. 1253 * A value of 0.0 results in zero gain (silence), and 1254 * a value of 1.0 means unity gain (signal unchanged). 1255 * The default value is 1.0 meaning unity gain. 1256 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 1257 * @param leftGain output gain for the left channel. 1258 * @param rightGain output gain for the right channel 1259 * @return error code or success, see {@link #SUCCESS}, 1260 * {@link #ERROR_INVALID_OPERATION} 1261 * @deprecated Applications should use {@link #setVolume} instead, as it 1262 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 1263 */ 1264 public int setStereoVolume(float leftGain, float rightGain) { 1265 if (isRestricted()) { 1266 return SUCCESS; 1267 } 1268 if (mState == STATE_UNINITIALIZED) { 1269 return ERROR_INVALID_OPERATION; 1270 } 1271 1272 leftGain = clampGainOrLevel(leftGain); 1273 rightGain = clampGainOrLevel(rightGain); 1274 1275 native_setVolume(leftGain, rightGain); 1276 1277 return SUCCESS; 1278 } 1279 1280 1281 /** 1282 * Sets the specified output gain value on all channels of this track. 1283 * <p>Gain values are clamped to the closed interval [0.0, max] where 1284 * max is the value of {@link #getMaxVolume}. 1285 * A value of 0.0 results in zero gain (silence), and 1286 * a value of 1.0 means unity gain (signal unchanged). 1287 * The default value is 1.0 meaning unity gain. 1288 * <p>This API is preferred over {@link #setStereoVolume}, as it 1289 * more gracefully scales down to mono, and up to multi-channel content beyond stereo. 1290 * <p>The word "volume" in the API name is historical; this is actually a linear gain. 1291 * @param gain output gain for all channels. 1292 * @return error code or success, see {@link #SUCCESS}, 1293 * {@link #ERROR_INVALID_OPERATION} 1294 */ 1295 public int setVolume(float gain) { 1296 return setStereoVolume(gain, gain); 1297 } 1298 1299 1300 /** 1301 * Sets the playback sample rate for this track. This sets the sampling rate at which 1302 * the audio data will be consumed and played back 1303 * (as set by the sampleRateInHz parameter in the 1304 * {@link #AudioTrack(int, int, int, int, int, int)} constructor), 1305 * not the original sampling rate of the 1306 * content. For example, setting it to half the sample rate of the content will cause the 1307 * playback to last twice as long, but will also result in a pitch shift down by one octave. 1308 * The valid sample rate range is from 1 Hz to twice the value returned by 1309 * {@link #getNativeOutputSampleRate(int)}. 1310 * Use {@link #setPlaybackSettings(PlaybackSettings)} for speed control. 1311 * @param sampleRateInHz the sample rate expressed in Hz 1312 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1313 * {@link #ERROR_INVALID_OPERATION} 1314 */ 1315 public int setPlaybackRate(int sampleRateInHz) { 1316 if (mState != STATE_INITIALIZED) { 1317 return ERROR_INVALID_OPERATION; 1318 } 1319 if (sampleRateInHz <= 0) { 1320 return ERROR_BAD_VALUE; 1321 } 1322 return native_set_playback_rate(sampleRateInHz); 1323 } 1324 1325 1326 /** 1327 * Sets the playback settings. 1328 * This method returns failure if it cannot apply the playback settings. 1329 * One possible cause is that the parameters for speed or pitch are out of range. 1330 * Another possible cause is that the <code>AudioTrack</code> is streaming 1331 * (see {@link #MODE_STREAM}) and the 1332 * buffer size is too small. For speeds greater than 1.0f, the <code>AudioTrack</code> buffer 1333 * on configuration must be larger than the speed multiplied by the minimum size 1334 * {@link #getMinBufferSize(int, int, int)}) to allow proper playback. 1335 * @param settings see {@link PlaybackSettings}. In particular, 1336 * speed, pitch, and audio mode should be set. 1337 * @throws IllegalArgumentException if the settings are invalid or not accepted. 1338 * @throws IllegalStateException if track is not initialized. 1339 */ 1340 public void setPlaybackSettings(@NonNull PlaybackSettings settings) { 1341 if (settings == null) { 1342 throw new IllegalArgumentException("settings is null"); 1343 } 1344 float[] floatArray; 1345 int[] intArray; 1346 try { 1347 floatArray = new float[] { 1348 settings.getSpeed(), 1349 settings.getPitch(), 1350 }; 1351 intArray = new int[] { 1352 settings.getAudioFallbackMode(), 1353 settings.getAudioStretchMode(), 1354 }; 1355 } catch (IllegalStateException e) { 1356 throw new IllegalArgumentException(e); 1357 } 1358 native_set_playback_settings(floatArray, intArray); 1359 } 1360 1361 1362 /** 1363 * Sets the position of the notification marker. At most one marker can be active. 1364 * @param markerInFrames marker position in wrapping frame units similar to 1365 * {@link #getPlaybackHeadPosition}, or zero to disable the marker. 1366 * To set a marker at a position which would appear as zero due to wraparound, 1367 * a workaround is to use a non-zero position near zero, such as -1 or 1. 1368 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1369 * {@link #ERROR_INVALID_OPERATION} 1370 */ 1371 public int setNotificationMarkerPosition(int markerInFrames) { 1372 if (mState == STATE_UNINITIALIZED) { 1373 return ERROR_INVALID_OPERATION; 1374 } 1375 return native_set_marker_pos(markerInFrames); 1376 } 1377 1378 1379 /** 1380 * Sets the period for the periodic notification event. 1381 * @param periodInFrames update period expressed in frames. 1382 * Zero period means no position updates. A negative period is not allowed. 1383 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_INVALID_OPERATION} 1384 */ 1385 public int setPositionNotificationPeriod(int periodInFrames) { 1386 if (mState == STATE_UNINITIALIZED) { 1387 return ERROR_INVALID_OPERATION; 1388 } 1389 return native_set_pos_update_period(periodInFrames); 1390 } 1391 1392 1393 /** 1394 * Sets the playback head position within the static buffer. 1395 * The track must be stopped or paused for the position to be changed, 1396 * and must use the {@link #MODE_STATIC} mode. 1397 * @param positionInFrames playback head position within buffer, expressed in frames. 1398 * Zero corresponds to start of buffer. 1399 * The position must not be greater than the buffer size in frames, or negative. 1400 * Though this method and {@link #getPlaybackHeadPosition()} have similar names, 1401 * the position values have different meanings. 1402 * <br> 1403 * If looping is currently enabled and the new position is greater than or equal to the 1404 * loop end marker, the behavior varies by API level: for API level 22 and above, 1405 * the looping is first disabled and then the position is set. 1406 * For earlier API levels, the behavior is unspecified. 1407 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1408 * {@link #ERROR_INVALID_OPERATION} 1409 */ 1410 public int setPlaybackHeadPosition(int positionInFrames) { 1411 if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED || 1412 getPlayState() == PLAYSTATE_PLAYING) { 1413 return ERROR_INVALID_OPERATION; 1414 } 1415 if (!(0 <= positionInFrames && positionInFrames <= mNativeBufferSizeInFrames)) { 1416 return ERROR_BAD_VALUE; 1417 } 1418 return native_set_position(positionInFrames); 1419 } 1420 1421 /** 1422 * Sets the loop points and the loop count. The loop can be infinite. 1423 * Similarly to setPlaybackHeadPosition, 1424 * the track must be stopped or paused for the loop points to be changed, 1425 * and must use the {@link #MODE_STATIC} mode. 1426 * @param startInFrames loop start marker expressed in frames. 1427 * Zero corresponds to start of buffer. 1428 * The start marker must not be greater than or equal to the buffer size in frames, or negative. 1429 * @param endInFrames loop end marker expressed in frames. 1430 * The total buffer size in frames corresponds to end of buffer. 1431 * The end marker must not be greater than the buffer size in frames. 1432 * For looping, the end marker must not be less than or equal to the start marker, 1433 * but to disable looping 1434 * it is permitted for start marker, end marker, and loop count to all be 0. 1435 * If any input parameters are out of range, this method returns {@link #ERROR_BAD_VALUE}. 1436 * If the loop period (endInFrames - startInFrames) is too small for the implementation to 1437 * support, 1438 * {@link #ERROR_BAD_VALUE} is returned. 1439 * The loop range is the interval [startInFrames, endInFrames). 1440 * <br> 1441 * For API level 22 and above, the position is left unchanged, 1442 * unless it is greater than or equal to the loop end marker, in which case 1443 * it is forced to the loop start marker. 1444 * For earlier API levels, the effect on position is unspecified. 1445 * @param loopCount the number of times the loop is looped; must be greater than or equal to -1. 1446 * A value of -1 means infinite looping, and 0 disables looping. 1447 * A value of positive N means to "loop" (go back) N times. For example, 1448 * a value of one means to play the region two times in total. 1449 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1450 * {@link #ERROR_INVALID_OPERATION} 1451 */ 1452 public int setLoopPoints(int startInFrames, int endInFrames, int loopCount) { 1453 if (mDataLoadMode == MODE_STREAM || mState == STATE_UNINITIALIZED || 1454 getPlayState() == PLAYSTATE_PLAYING) { 1455 return ERROR_INVALID_OPERATION; 1456 } 1457 if (loopCount == 0) { 1458 ; // explicitly allowed as an exception to the loop region range check 1459 } else if (!(0 <= startInFrames && startInFrames < mNativeBufferSizeInFrames && 1460 startInFrames < endInFrames && endInFrames <= mNativeBufferSizeInFrames)) { 1461 return ERROR_BAD_VALUE; 1462 } 1463 return native_set_loop(startInFrames, endInFrames, loopCount); 1464 } 1465 1466 /** 1467 * Sets the initialization state of the instance. This method was originally intended to be used 1468 * in an AudioTrack subclass constructor to set a subclass-specific post-initialization state. 1469 * However, subclasses of AudioTrack are no longer recommended, so this method is obsolete. 1470 * @param state the state of the AudioTrack instance 1471 * @deprecated Only accessible by subclasses, which are not recommended for AudioTrack. 1472 */ 1473 @Deprecated 1474 protected void setState(int state) { 1475 mState = state; 1476 } 1477 1478 1479 //--------------------------------------------------------- 1480 // Transport control methods 1481 //-------------------- 1482 /** 1483 * Starts playing an AudioTrack. 1484 * If track's creation mode is {@link #MODE_STATIC}, you must have called one of 1485 * the {@link #write(byte[], int, int)}, {@link #write(short[], int, int)}, 1486 * or {@link #write(float[], int, int, int)} methods. 1487 * If the mode is {@link #MODE_STREAM}, you can optionally prime the 1488 * output buffer by writing up to bufferSizeInBytes (from constructor) before starting. 1489 * This priming will avoid an immediate underrun, but is not required. 1490 * 1491 * @throws IllegalStateException 1492 */ 1493 public void play() 1494 throws IllegalStateException { 1495 if (mState != STATE_INITIALIZED) { 1496 throw new IllegalStateException("play() called on uninitialized AudioTrack."); 1497 } 1498 if (isRestricted()) { 1499 setVolume(0); 1500 } 1501 synchronized(mPlayStateLock) { 1502 native_start(); 1503 mPlayState = PLAYSTATE_PLAYING; 1504 } 1505 } 1506 1507 private boolean isRestricted() { 1508 if ((mAttributes.getFlags() & AudioAttributes.FLAG_BYPASS_INTERRUPTION_POLICY) != 0) { 1509 return false; 1510 } 1511 try { 1512 final int usage = AudioAttributes.usageForLegacyStreamType(mStreamType); 1513 final int mode = mAppOps.checkAudioOperation(AppOpsManager.OP_PLAY_AUDIO, usage, 1514 Process.myUid(), ActivityThread.currentPackageName()); 1515 return mode != AppOpsManager.MODE_ALLOWED; 1516 } catch (RemoteException e) { 1517 return false; 1518 } 1519 } 1520 1521 /** 1522 * Stops playing the audio data. 1523 * When used on an instance created in {@link #MODE_STREAM} mode, audio will stop playing 1524 * after the last buffer that was written has been played. For an immediate stop, use 1525 * {@link #pause()}, followed by {@link #flush()} to discard audio data that hasn't been played 1526 * back yet. 1527 * @throws IllegalStateException 1528 */ 1529 public void stop() 1530 throws IllegalStateException { 1531 if (mState != STATE_INITIALIZED) { 1532 throw new IllegalStateException("stop() called on uninitialized AudioTrack."); 1533 } 1534 1535 // stop playing 1536 synchronized(mPlayStateLock) { 1537 native_stop(); 1538 mPlayState = PLAYSTATE_STOPPED; 1539 mAvSyncHeader = null; 1540 mAvSyncBytesRemaining = 0; 1541 } 1542 } 1543 1544 /** 1545 * Pauses the playback of the audio data. Data that has not been played 1546 * back will not be discarded. Subsequent calls to {@link #play} will play 1547 * this data back. See {@link #flush()} to discard this data. 1548 * 1549 * @throws IllegalStateException 1550 */ 1551 public void pause() 1552 throws IllegalStateException { 1553 if (mState != STATE_INITIALIZED) { 1554 throw new IllegalStateException("pause() called on uninitialized AudioTrack."); 1555 } 1556 //logd("pause()"); 1557 1558 // pause playback 1559 synchronized(mPlayStateLock) { 1560 native_pause(); 1561 mPlayState = PLAYSTATE_PAUSED; 1562 } 1563 } 1564 1565 1566 //--------------------------------------------------------- 1567 // Audio data supply 1568 //-------------------- 1569 1570 /** 1571 * Flushes the audio data currently queued for playback. Any data that has 1572 * been written but not yet presented will be discarded. No-op if not stopped or paused, 1573 * or if the track's creation mode is not {@link #MODE_STREAM}. 1574 * <BR> Note that although data written but not yet presented is discarded, there is no 1575 * guarantee that all of the buffer space formerly used by that data 1576 * is available for a subsequent write. 1577 * For example, a call to {@link #write(byte[], int, int)} with <code>sizeInBytes</code> 1578 * less than or equal to the total buffer size 1579 * may return a short actual transfer count. 1580 */ 1581 public void flush() { 1582 if (mState == STATE_INITIALIZED) { 1583 // flush the data in native layer 1584 native_flush(); 1585 mAvSyncHeader = null; 1586 mAvSyncBytesRemaining = 0; 1587 } 1588 1589 } 1590 1591 /** 1592 * Writes the audio data to the audio sink for playback (streaming mode), 1593 * or copies audio data for later playback (static buffer mode). 1594 * The format specified in the AudioTrack constructor should be 1595 * {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array. 1596 * In streaming mode, will block until all data has been written to the audio sink. 1597 * In static buffer mode, copies the data to the buffer starting at offset 0. 1598 * Note that the actual playback of this data might occur after this function 1599 * returns. This function is thread safe with respect to {@link #stop} calls, 1600 * in which case all of the specified data might not be written to the audio sink. 1601 * 1602 * @param audioData the array that holds the data to play. 1603 * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 1604 * starts. 1605 * @param sizeInBytes the number of bytes to read in audioData after the offset. 1606 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1607 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1608 * the parameters don't resolve to valid data and indexes, or 1609 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1610 * needs to be recreated. 1611 */ 1612 public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes) { 1613 return write(audioData, offsetInBytes, sizeInBytes, WRITE_BLOCKING); 1614 } 1615 1616 /** 1617 * Writes the audio data to the audio sink for playback (streaming mode), 1618 * or copies audio data for later playback (static buffer mode). 1619 * The format specified in the AudioTrack constructor should be 1620 * {@link AudioFormat#ENCODING_PCM_8BIT} to correspond to the data in the array. 1621 * In streaming mode, will block until all data has been written to the audio sink. 1622 * In static buffer mode, copies the data to the buffer starting at offset 0. 1623 * Note that the actual playback of this data might occur after this function 1624 * returns. This function is thread safe with respect to {@link #stop} calls, 1625 * in which case all of the specified data might not be written to the audio sink. 1626 * 1627 * @param audioData the array that holds the data to play. 1628 * @param offsetInBytes the offset expressed in bytes in audioData where the data to play 1629 * starts. 1630 * @param sizeInBytes the number of bytes to read in audioData after the offset. 1631 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1632 * effect in static mode. 1633 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1634 * to the audio sink. 1635 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1636 * queuing as much audio data for playback as possible without blocking. 1637 * @return the number of bytes that were written or {@link #ERROR_INVALID_OPERATION} 1638 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1639 * the parameters don't resolve to valid data and indexes, or 1640 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1641 * needs to be recreated. 1642 */ 1643 public int write(@NonNull byte[] audioData, int offsetInBytes, int sizeInBytes, 1644 @WriteMode int writeMode) { 1645 1646 if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1647 return ERROR_INVALID_OPERATION; 1648 } 1649 1650 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1651 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1652 return ERROR_BAD_VALUE; 1653 } 1654 1655 if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0) 1656 || (offsetInBytes + sizeInBytes < 0) // detect integer overflow 1657 || (offsetInBytes + sizeInBytes > audioData.length)) { 1658 return ERROR_BAD_VALUE; 1659 } 1660 1661 int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat, 1662 writeMode == WRITE_BLOCKING); 1663 1664 if ((mDataLoadMode == MODE_STATIC) 1665 && (mState == STATE_NO_STATIC_DATA) 1666 && (ret > 0)) { 1667 // benign race with respect to other APIs that read mState 1668 mState = STATE_INITIALIZED; 1669 } 1670 1671 return ret; 1672 } 1673 1674 /** 1675 * Writes the audio data to the audio sink for playback (streaming mode), 1676 * or copies audio data for later playback (static buffer mode). 1677 * The format specified in the AudioTrack constructor should be 1678 * {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array. 1679 * In streaming mode, will block until all data has been written to the audio sink. 1680 * In static buffer mode, copies the data to the buffer starting at offset 0. 1681 * Note that the actual playback of this data might occur after this function 1682 * returns. This function is thread safe with respect to {@link #stop} calls, 1683 * in which case all of the specified data might not be written to the audio sink. 1684 * 1685 * @param audioData the array that holds the data to play. 1686 * @param offsetInShorts the offset expressed in shorts in audioData where the data to play 1687 * starts. 1688 * @param sizeInShorts the number of shorts to read in audioData after the offset. 1689 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1690 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1691 * the parameters don't resolve to valid data and indexes, or 1692 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1693 * needs to be recreated. 1694 */ 1695 public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts) { 1696 return write(audioData, offsetInShorts, sizeInShorts, WRITE_BLOCKING); 1697 } 1698 1699 /** 1700 * Writes the audio data to the audio sink for playback (streaming mode), 1701 * or copies audio data for later playback (static buffer mode). 1702 * The format specified in the AudioTrack constructor should be 1703 * {@link AudioFormat#ENCODING_PCM_16BIT} to correspond to the data in the array. 1704 * In streaming mode, will block until all data has been written to the audio sink. 1705 * In static buffer mode, copies the data to the buffer starting at offset 0. 1706 * Note that the actual playback of this data might occur after this function 1707 * returns. This function is thread safe with respect to {@link #stop} calls, 1708 * in which case all of the specified data might not be written to the audio sink. 1709 * 1710 * @param audioData the array that holds the data to play. 1711 * @param offsetInShorts the offset expressed in shorts in audioData where the data to play 1712 * starts. 1713 * @param sizeInShorts the number of shorts to read in audioData after the offset. 1714 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1715 * effect in static mode. 1716 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1717 * to the audio sink. 1718 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1719 * queuing as much audio data for playback as possible without blocking. 1720 * @return the number of shorts that were written or {@link #ERROR_INVALID_OPERATION} 1721 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1722 * the parameters don't resolve to valid data and indexes, or 1723 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1724 * needs to be recreated. 1725 */ 1726 public int write(@NonNull short[] audioData, int offsetInShorts, int sizeInShorts, 1727 @WriteMode int writeMode) { 1728 1729 if (mState == STATE_UNINITIALIZED || mAudioFormat == AudioFormat.ENCODING_PCM_FLOAT) { 1730 return ERROR_INVALID_OPERATION; 1731 } 1732 1733 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1734 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1735 return ERROR_BAD_VALUE; 1736 } 1737 1738 if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0) 1739 || (offsetInShorts + sizeInShorts < 0) // detect integer overflow 1740 || (offsetInShorts + sizeInShorts > audioData.length)) { 1741 return ERROR_BAD_VALUE; 1742 } 1743 1744 int ret = native_write_short(audioData, offsetInShorts, sizeInShorts, mAudioFormat, 1745 writeMode == WRITE_BLOCKING); 1746 1747 if ((mDataLoadMode == MODE_STATIC) 1748 && (mState == STATE_NO_STATIC_DATA) 1749 && (ret > 0)) { 1750 // benign race with respect to other APIs that read mState 1751 mState = STATE_INITIALIZED; 1752 } 1753 1754 return ret; 1755 } 1756 1757 /** 1758 * Writes the audio data to the audio sink for playback (streaming mode), 1759 * or copies audio data for later playback (static buffer mode). 1760 * The format specified in the AudioTrack constructor should be 1761 * {@link AudioFormat#ENCODING_PCM_FLOAT} to correspond to the data in the array. 1762 * In static buffer mode, copies the data to the buffer starting at offset 0, 1763 * and the write mode is ignored. 1764 * In streaming mode, the blocking behavior will depend on the write mode. 1765 * <p> 1766 * Note that the actual playback of this data might occur after this function 1767 * returns. This function is thread safe with respect to {@link #stop} calls, 1768 * in which case all of the specified data might not be written to the audio sink. 1769 * <p> 1770 * @param audioData the array that holds the data to play. 1771 * The implementation does not clip for sample values within the nominal range 1772 * [-1.0f, 1.0f], provided that all gains in the audio pipeline are 1773 * less than or equal to unity (1.0f), and in the absence of post-processing effects 1774 * that could add energy, such as reverb. For the convenience of applications 1775 * that compute samples using filters with non-unity gain, 1776 * sample values +3 dB beyond the nominal range are permitted. 1777 * However such values may eventually be limited or clipped, depending on various gains 1778 * and later processing in the audio path. Therefore applications are encouraged 1779 * to provide samples values within the nominal range. 1780 * @param offsetInFloats the offset, expressed as a number of floats, 1781 * in audioData where the data to play starts. 1782 * @param sizeInFloats the number of floats to read in audioData after the offset. 1783 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1784 * effect in static mode. 1785 * <br>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1786 * to the audio sink. 1787 * <br>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1788 * queuing as much audio data for playback as possible without blocking. 1789 * @return the number of floats that were written, or {@link #ERROR_INVALID_OPERATION} 1790 * if the object wasn't properly initialized, or {@link #ERROR_BAD_VALUE} if 1791 * the parameters don't resolve to valid data and indexes, or 1792 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1793 * needs to be recreated. 1794 */ 1795 public int write(@NonNull float[] audioData, int offsetInFloats, int sizeInFloats, 1796 @WriteMode int writeMode) { 1797 1798 if (mState == STATE_UNINITIALIZED) { 1799 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1800 return ERROR_INVALID_OPERATION; 1801 } 1802 1803 if (mAudioFormat != AudioFormat.ENCODING_PCM_FLOAT) { 1804 Log.e(TAG, "AudioTrack.write(float[] ...) requires format ENCODING_PCM_FLOAT"); 1805 return ERROR_INVALID_OPERATION; 1806 } 1807 1808 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1809 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1810 return ERROR_BAD_VALUE; 1811 } 1812 1813 if ( (audioData == null) || (offsetInFloats < 0 ) || (sizeInFloats < 0) 1814 || (offsetInFloats + sizeInFloats < 0) // detect integer overflow 1815 || (offsetInFloats + sizeInFloats > audioData.length)) { 1816 Log.e(TAG, "AudioTrack.write() called with invalid array, offset, or size"); 1817 return ERROR_BAD_VALUE; 1818 } 1819 1820 int ret = native_write_float(audioData, offsetInFloats, sizeInFloats, mAudioFormat, 1821 writeMode == WRITE_BLOCKING); 1822 1823 if ((mDataLoadMode == MODE_STATIC) 1824 && (mState == STATE_NO_STATIC_DATA) 1825 && (ret > 0)) { 1826 // benign race with respect to other APIs that read mState 1827 mState = STATE_INITIALIZED; 1828 } 1829 1830 return ret; 1831 } 1832 1833 1834 /** 1835 * Writes the audio data to the audio sink for playback (streaming mode), 1836 * or copies audio data for later playback (static buffer mode). 1837 * In static buffer mode, copies the data to the buffer starting at its 0 offset, and the write 1838 * mode is ignored. 1839 * In streaming mode, the blocking behavior will depend on the write mode. 1840 * @param audioData the buffer that holds the data to play, starting at the position reported 1841 * by <code>audioData.position()</code>. 1842 * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will 1843 * have been advanced to reflect the amount of data that was successfully written to 1844 * the AudioTrack. 1845 * @param sizeInBytes number of bytes to write. 1846 * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. 1847 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. It has no 1848 * effect in static mode. 1849 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1850 * to the audio sink. 1851 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1852 * queuing as much audio data for playback as possible without blocking. 1853 * @return 0 or a positive number of bytes that were written, or 1854 * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}, or 1855 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1856 * needs to be recreated. 1857 */ 1858 public int write(@NonNull ByteBuffer audioData, int sizeInBytes, 1859 @WriteMode int writeMode) { 1860 1861 if (mState == STATE_UNINITIALIZED) { 1862 Log.e(TAG, "AudioTrack.write() called in invalid state STATE_UNINITIALIZED"); 1863 return ERROR_INVALID_OPERATION; 1864 } 1865 1866 if ((writeMode != WRITE_BLOCKING) && (writeMode != WRITE_NON_BLOCKING)) { 1867 Log.e(TAG, "AudioTrack.write() called with invalid blocking mode"); 1868 return ERROR_BAD_VALUE; 1869 } 1870 1871 if ( (audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { 1872 Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); 1873 return ERROR_BAD_VALUE; 1874 } 1875 1876 int ret = 0; 1877 if (audioData.isDirect()) { 1878 ret = native_write_native_bytes(audioData, 1879 audioData.position(), sizeInBytes, mAudioFormat, 1880 writeMode == WRITE_BLOCKING); 1881 } else { 1882 ret = native_write_byte(NioUtils.unsafeArray(audioData), 1883 NioUtils.unsafeArrayOffset(audioData) + audioData.position(), 1884 sizeInBytes, mAudioFormat, 1885 writeMode == WRITE_BLOCKING); 1886 } 1887 1888 if ((mDataLoadMode == MODE_STATIC) 1889 && (mState == STATE_NO_STATIC_DATA) 1890 && (ret > 0)) { 1891 // benign race with respect to other APIs that read mState 1892 mState = STATE_INITIALIZED; 1893 } 1894 1895 if (ret > 0) { 1896 audioData.position(audioData.position() + ret); 1897 } 1898 1899 return ret; 1900 } 1901 1902 /** 1903 * Writes the audio data to the audio sink for playback (streaming mode) on a HW_AV_SYNC track. 1904 * In streaming mode, the blocking behavior will depend on the write mode. 1905 * @param audioData the buffer that holds the data to play, starting at the position reported 1906 * by <code>audioData.position()</code>. 1907 * <BR>Note that upon return, the buffer position (<code>audioData.position()</code>) will 1908 * have been advanced to reflect the amount of data that was successfully written to 1909 * the AudioTrack. 1910 * @param sizeInBytes number of bytes to write. 1911 * <BR>Note this may differ from <code>audioData.remaining()</code>, but cannot exceed it. 1912 * @param writeMode one of {@link #WRITE_BLOCKING}, {@link #WRITE_NON_BLOCKING}. 1913 * <BR>With {@link #WRITE_BLOCKING}, the write will block until all data has been written 1914 * to the audio sink. 1915 * <BR>With {@link #WRITE_NON_BLOCKING}, the write will return immediately after 1916 * queuing as much audio data for playback as possible without blocking. 1917 * @param timestamp The timestamp of the first decodable audio frame in the provided audioData. 1918 * @return 0 or a positive number of bytes that were written, or 1919 * {@link #ERROR_BAD_VALUE}, {@link #ERROR_INVALID_OPERATION}, or 1920 * {@link AudioManager#ERROR_DEAD_OBJECT} if the AudioTrack is not valid anymore and 1921 * needs to be recreated. 1922 */ 1923 public int write(ByteBuffer audioData, int sizeInBytes, 1924 @WriteMode int writeMode, long timestamp) { 1925 1926 if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) == 0) { 1927 Log.d(TAG, "AudioTrack.write() called on a regular AudioTrack. Ignoring pts..."); 1928 return write(audioData, sizeInBytes, writeMode); 1929 } 1930 1931 if ((audioData == null) || (sizeInBytes < 0) || (sizeInBytes > audioData.remaining())) { 1932 Log.e(TAG, "AudioTrack.write() called with invalid size (" + sizeInBytes + ") value"); 1933 return ERROR_BAD_VALUE; 1934 } 1935 1936 // create timestamp header if none exists 1937 if (mAvSyncHeader == null) { 1938 mAvSyncHeader = ByteBuffer.allocate(16); 1939 mAvSyncHeader.order(ByteOrder.BIG_ENDIAN); 1940 mAvSyncHeader.putInt(0x55550001); 1941 mAvSyncHeader.putInt(sizeInBytes); 1942 mAvSyncHeader.putLong(timestamp); 1943 mAvSyncHeader.position(0); 1944 mAvSyncBytesRemaining = sizeInBytes; 1945 } 1946 1947 // write timestamp header if not completely written already 1948 int ret = 0; 1949 if (mAvSyncHeader.remaining() != 0) { 1950 ret = write(mAvSyncHeader, mAvSyncHeader.remaining(), writeMode); 1951 if (ret < 0) { 1952 Log.e(TAG, "AudioTrack.write() could not write timestamp header!"); 1953 mAvSyncHeader = null; 1954 mAvSyncBytesRemaining = 0; 1955 return ret; 1956 } 1957 if (mAvSyncHeader.remaining() > 0) { 1958 Log.v(TAG, "AudioTrack.write() partial timestamp header written."); 1959 return 0; 1960 } 1961 } 1962 1963 // write audio data 1964 int sizeToWrite = Math.min(mAvSyncBytesRemaining, sizeInBytes); 1965 ret = write(audioData, sizeToWrite, writeMode); 1966 if (ret < 0) { 1967 Log.e(TAG, "AudioTrack.write() could not write audio data!"); 1968 mAvSyncHeader = null; 1969 mAvSyncBytesRemaining = 0; 1970 return ret; 1971 } 1972 1973 mAvSyncBytesRemaining -= ret; 1974 if (mAvSyncBytesRemaining == 0) { 1975 mAvSyncHeader = null; 1976 } 1977 1978 return ret; 1979 } 1980 1981 1982 /** 1983 * Sets the playback head position within the static buffer to zero, 1984 * that is it rewinds to start of static buffer. 1985 * The track must be stopped or paused, and 1986 * the track's creation mode must be {@link #MODE_STATIC}. 1987 * <p> 1988 * For API level 22 and above, also resets the value returned by 1989 * {@link #getPlaybackHeadPosition()} to zero. 1990 * For earlier API levels, the reset behavior is unspecified. 1991 * <p> 1992 * {@link #setPlaybackHeadPosition(int)} to zero 1993 * is recommended instead when the reset of {@link #getPlaybackHeadPosition} is not needed. 1994 * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE}, 1995 * {@link #ERROR_INVALID_OPERATION} 1996 */ 1997 public int reloadStaticData() { 1998 if (mDataLoadMode == MODE_STREAM || mState != STATE_INITIALIZED) { 1999 return ERROR_INVALID_OPERATION; 2000 } 2001 return native_reload_static(); 2002 } 2003 2004 //-------------------------------------------------------------------------- 2005 // Audio effects management 2006 //-------------------- 2007 2008 /** 2009 * Attaches an auxiliary effect to the audio track. A typical auxiliary 2010 * effect is a reverberation effect which can be applied on any sound source 2011 * that directs a certain amount of its energy to this effect. This amount 2012 * is defined by setAuxEffectSendLevel(). 2013 * {@see #setAuxEffectSendLevel(float)}. 2014 * <p>After creating an auxiliary effect (e.g. 2015 * {@link android.media.audiofx.EnvironmentalReverb}), retrieve its ID with 2016 * {@link android.media.audiofx.AudioEffect#getId()} and use it when calling 2017 * this method to attach the audio track to the effect. 2018 * <p>To detach the effect from the audio track, call this method with a 2019 * null effect id. 2020 * 2021 * @param effectId system wide unique id of the effect to attach 2022 * @return error code or success, see {@link #SUCCESS}, 2023 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR_BAD_VALUE} 2024 */ 2025 public int attachAuxEffect(int effectId) { 2026 if (mState == STATE_UNINITIALIZED) { 2027 return ERROR_INVALID_OPERATION; 2028 } 2029 return native_attachAuxEffect(effectId); 2030 } 2031 2032 /** 2033 * Sets the send level of the audio track to the attached auxiliary effect 2034 * {@link #attachAuxEffect(int)}. Effect levels 2035 * are clamped to the closed interval [0.0, max] where 2036 * max is the value of {@link #getMaxVolume}. 2037 * A value of 0.0 results in no effect, and a value of 1.0 is full send. 2038 * <p>By default the send level is 0.0f, so even if an effect is attached to the player 2039 * this method must be called for the effect to be applied. 2040 * <p>Note that the passed level value is a linear scalar. UI controls should be scaled 2041 * logarithmically: the gain applied by audio framework ranges from -72dB to at least 0dB, 2042 * so an appropriate conversion from linear UI input x to level is: 2043 * x == 0 -> level = 0 2044 * 0 < x <= R -> level = 10^(72*(x-R)/20/R) 2045 * 2046 * @param level linear send level 2047 * @return error code or success, see {@link #SUCCESS}, 2048 * {@link #ERROR_INVALID_OPERATION}, {@link #ERROR} 2049 */ 2050 public int setAuxEffectSendLevel(float level) { 2051 if (isRestricted()) { 2052 return SUCCESS; 2053 } 2054 if (mState == STATE_UNINITIALIZED) { 2055 return ERROR_INVALID_OPERATION; 2056 } 2057 level = clampGainOrLevel(level); 2058 int err = native_setAuxEffectSendLevel(level); 2059 return err == 0 ? SUCCESS : ERROR; 2060 } 2061 2062 //-------------------------------------------------------------------------- 2063 // Explicit Routing 2064 //-------------------- 2065 private AudioDeviceInfo mPreferredDevice = null; 2066 2067 /** 2068 * Specifies an audio device (via an {@link AudioDeviceInfo} object) to route 2069 * the output from this AudioTrack. 2070 * @param deviceInfo The {@link AudioDeviceInfo} specifying the audio sink. 2071 * If deviceInfo is null, default routing is restored. 2072 * @return true if succesful, false if the specified {@link AudioDeviceInfo} is non-null and 2073 * does not correspond to a valid audio output device. 2074 */ 2075 public boolean setPreferredOutputDevice(AudioDeviceInfo deviceInfo) { 2076 // Do some validation.... 2077 if (deviceInfo != null && !deviceInfo.isSink()) { 2078 return false; 2079 } 2080 2081 mPreferredDevice = deviceInfo; 2082 int preferredDeviceId = mPreferredDevice != null ? deviceInfo.getId() : 0; 2083 2084 return native_setOutputDevice(preferredDeviceId); 2085 } 2086 2087 /** 2088 * Returns the selected output specified by {@link #setPreferredOutputDevice}. Note that this 2089 * is not guaranteed to correspond to the actual device being used for playback. 2090 */ 2091 public AudioDeviceInfo getPreferredOutputDevice() { 2092 return mPreferredDevice; 2093 } 2094 2095 //-------------------------------------------------------------------------- 2096 // (Re)Routing Info 2097 //-------------------- 2098 /** 2099 * Returns an {@link AudioDeviceInfo} identifying the current routing of this AudioTrack. 2100 */ 2101 public AudioDeviceInfo getRoutedDevice() { 2102 return null; 2103 } 2104 2105 /** 2106 * The message sent to apps when the routing of this AudioTrack changes if they provide 2107 * a {#link Handler} object to addOnAudioTrackRoutingListener(). 2108 */ 2109 private ArrayMap<OnAudioTrackRoutingListener, NativeRoutingEventHandlerDelegate> 2110 mRoutingChangeListeners = 2111 new ArrayMap<OnAudioTrackRoutingListener, NativeRoutingEventHandlerDelegate>(); 2112 2113 /** 2114 * Adds an {@link OnAudioTrackRoutingListener} to receive notifications of routing changes 2115 * on this AudioTrack. 2116 */ 2117 public void addOnAudioTrackRoutingListener(OnAudioTrackRoutingListener listener, 2118 android.os.Handler handler) { 2119 if (listener != null && !mRoutingChangeListeners.containsKey(listener)) { 2120 synchronized (mRoutingChangeListeners) { 2121 mRoutingChangeListeners.put( 2122 listener, new NativeRoutingEventHandlerDelegate(this, listener, handler)); 2123 } 2124 } 2125 } 2126 2127 /** 2128 * Removes an {@link OnAudioTrackRoutingListener} which has been previously added 2129 * to receive notifications of changes to the set of connected audio devices. 2130 */ 2131 public void removeOnAudioTrackRoutingListener(OnAudioTrackRoutingListener listener) { 2132 synchronized (mRoutingChangeListeners) { 2133 if (mRoutingChangeListeners.containsKey(listener)) { 2134 mRoutingChangeListeners.remove(listener); 2135 } 2136 } 2137 } 2138 2139 /** 2140 * Sends device list change notification to all listeners. 2141 */ 2142 private void broadcastRoutingChange() { 2143 Collection<NativeRoutingEventHandlerDelegate> values; 2144 synchronized (mRoutingChangeListeners) { 2145 values = mRoutingChangeListeners.values(); 2146 } 2147 for(NativeRoutingEventHandlerDelegate delegate : values) { 2148 Handler handler = delegate.getHandler(); 2149 if (handler != null) { 2150 handler.sendEmptyMessage(NATIVE_EVENT_ROUTING_CHANGE); 2151 } 2152 } 2153 } 2154 2155 //--------------------------------------------------------- 2156 // Interface definitions 2157 //-------------------- 2158 /** 2159 * Interface definition for a callback to be invoked when the playback head position of 2160 * an AudioTrack has reached a notification marker or has increased by a certain period. 2161 */ 2162 public interface OnPlaybackPositionUpdateListener { 2163 /** 2164 * Called on the listener to notify it that the previously set marker has been reached 2165 * by the playback head. 2166 */ 2167 void onMarkerReached(AudioTrack track); 2168 2169 /** 2170 * Called on the listener to periodically notify it that the playback head has reached 2171 * a multiple of the notification period. 2172 */ 2173 void onPeriodicNotification(AudioTrack track); 2174 } 2175 2176 //--------------------------------------------------------- 2177 // Inner classes 2178 //-------------------- 2179 /** 2180 * Helper class to handle the forwarding of native events to the appropriate listener 2181 * (potentially) handled in a different thread 2182 */ 2183 private class NativePositionEventHandlerDelegate { 2184 private final Handler mHandler; 2185 2186 NativePositionEventHandlerDelegate(final AudioTrack track, 2187 final OnPlaybackPositionUpdateListener listener, 2188 Handler handler) { 2189 // find the looper for our new event handler 2190 Looper looper; 2191 if (handler != null) { 2192 looper = handler.getLooper(); 2193 } else { 2194 // no given handler, use the looper the AudioTrack was created in 2195 looper = mInitializationLooper; 2196 } 2197 2198 // construct the event handler with this looper 2199 if (looper != null) { 2200 // implement the event handler delegate 2201 mHandler = new Handler(looper) { 2202 @Override 2203 public void handleMessage(Message msg) { 2204 if (track == null) { 2205 return; 2206 } 2207 switch(msg.what) { 2208 case NATIVE_EVENT_MARKER: 2209 if (listener != null) { 2210 listener.onMarkerReached(track); 2211 } 2212 break; 2213 case NATIVE_EVENT_NEW_POS: 2214 if (listener != null) { 2215 listener.onPeriodicNotification(track); 2216 } 2217 break; 2218 default: 2219 loge("Unknown native event type: " + msg.what); 2220 break; 2221 } 2222 } 2223 }; 2224 } else { 2225 mHandler = null; 2226 } 2227 } 2228 2229 Handler getHandler() { 2230 return mHandler; 2231 } 2232 } 2233 2234 /** 2235 * Helper class to handle the forwarding of native events to the appropriate listener 2236 * (potentially) handled in a different thread 2237 */ 2238 private class NativeRoutingEventHandlerDelegate { 2239 private final Handler mHandler; 2240 2241 NativeRoutingEventHandlerDelegate(final AudioTrack track, 2242 final OnAudioTrackRoutingListener listener, 2243 Handler handler) { 2244 // find the looper for our new event handler 2245 Looper looper; 2246 if (handler != null) { 2247 looper = handler.getLooper(); 2248 } else { 2249 // no given handler, use the looper the AudioTrack was created in 2250 looper = mInitializationLooper; 2251 } 2252 2253 // construct the event handler with this looper 2254 if (looper != null) { 2255 // implement the event handler delegate 2256 mHandler = new Handler(looper) { 2257 @Override 2258 public void handleMessage(Message msg) { 2259 if (track == null) { 2260 return; 2261 } 2262 switch(msg.what) { 2263 case NATIVE_EVENT_ROUTING_CHANGE: 2264 if (listener != null) { 2265 listener.onAudioTrackRouting(track); 2266 } 2267 break; 2268 default: 2269 loge("Unknown native event type: " + msg.what); 2270 break; 2271 } 2272 } 2273 }; 2274 } else { 2275 mHandler = null; 2276 } 2277 } 2278 2279 Handler getHandler() { 2280 return mHandler; 2281 } 2282 } 2283 2284 //--------------------------------------------------------- 2285 // Java methods called from the native side 2286 //-------------------- 2287 @SuppressWarnings("unused") 2288 private static void postEventFromNative(Object audiotrack_ref, 2289 int what, int arg1, int arg2, Object obj) { 2290 //logd("Event posted from the native side: event="+ what + " args="+ arg1+" "+arg2); 2291 AudioTrack track = (AudioTrack)((WeakReference)audiotrack_ref).get(); 2292 if (track == null) { 2293 return; 2294 } 2295 2296 NativePositionEventHandlerDelegate delegate = track.mEventHandlerDelegate; 2297 if (delegate != null) { 2298 Handler handler = delegate.getHandler(); 2299 if (handler != null) { 2300 Message m = handler.obtainMessage(what, arg1, arg2, obj); 2301 handler.sendMessage(m); 2302 } 2303 } 2304 2305 } 2306 2307 2308 //--------------------------------------------------------- 2309 // Native methods called from the Java side 2310 //-------------------- 2311 2312 // post-condition: mStreamType is overwritten with a value 2313 // that reflects the audio attributes (e.g. an AudioAttributes object with a usage of 2314 // AudioAttributes.USAGE_MEDIA will map to AudioManager.STREAM_MUSIC 2315 private native final int native_setup(Object /*WeakReference<AudioTrack>*/ audiotrack_this, 2316 Object /*AudioAttributes*/ attributes, 2317 int sampleRate, int channelMask, int channelIndexMask, int audioFormat, 2318 int buffSizeInBytes, int mode, int[] sessionId); 2319 2320 private native final void native_finalize(); 2321 2322 private native final void native_release(); 2323 2324 private native final void native_start(); 2325 2326 private native final void native_stop(); 2327 2328 private native final void native_pause(); 2329 2330 private native final void native_flush(); 2331 2332 private native final int native_write_byte(byte[] audioData, 2333 int offsetInBytes, int sizeInBytes, int format, 2334 boolean isBlocking); 2335 2336 private native final int native_write_short(short[] audioData, 2337 int offsetInShorts, int sizeInShorts, int format, 2338 boolean isBlocking); 2339 2340 private native final int native_write_float(float[] audioData, 2341 int offsetInFloats, int sizeInFloats, int format, 2342 boolean isBlocking); 2343 2344 private native final int native_write_native_bytes(Object audioData, 2345 int positionInBytes, int sizeInBytes, int format, boolean blocking); 2346 2347 private native final int native_reload_static(); 2348 2349 private native final int native_get_native_frame_count(); 2350 2351 private native final void native_setVolume(float leftVolume, float rightVolume); 2352 2353 private native final int native_set_playback_rate(int sampleRateInHz); 2354 private native final int native_get_playback_rate(); 2355 2356 // floatArray must be a non-null array of length >= 2 2357 // [0] is speed 2358 // [1] is pitch 2359 // intArray must be a non-null array of length >= 2 2360 // [0] is audio fallback mode 2361 // [1] is audio stretch mode 2362 private native final void native_set_playback_settings(float[] floatArray, int[] intArray); 2363 private native final void native_get_playback_settings(float[] floatArray, int[] intArray); 2364 2365 private native final int native_set_marker_pos(int marker); 2366 private native final int native_get_marker_pos(); 2367 2368 private native final int native_set_pos_update_period(int updatePeriod); 2369 private native final int native_get_pos_update_period(); 2370 2371 private native final int native_set_position(int position); 2372 private native final int native_get_position(); 2373 2374 private native final int native_get_latency(); 2375 2376 // longArray must be a non-null array of length >= 2 2377 // [0] is assigned the frame position 2378 // [1] is assigned the time in CLOCK_MONOTONIC nanoseconds 2379 private native final int native_get_timestamp(long[] longArray); 2380 2381 private native final int native_set_loop(int start, int end, int loopCount); 2382 2383 static private native final int native_get_output_sample_rate(int streamType); 2384 static private native final int native_get_min_buff_size( 2385 int sampleRateInHz, int channelConfig, int audioFormat); 2386 2387 private native final int native_attachAuxEffect(int effectId); 2388 private native final int native_setAuxEffectSendLevel(float level); 2389 2390 private native final boolean native_setOutputDevice(int deviceId); 2391 2392 //--------------------------------------------------------- 2393 // Utility methods 2394 //------------------ 2395 2396 private static void logd(String msg) { 2397 Log.d(TAG, msg); 2398 } 2399 2400 private static void loge(String msg) { 2401 Log.e(TAG, msg); 2402 } 2403} 2404